INVITE and hangup in the middle of a call
Hi guys, Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm. The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light. send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid
;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid>;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
Thanks Michael, I'll see if we can do that! So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE. On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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The session expires? On 20 May 2015 04:44, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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session timer On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com <javascript:_e(%7B%7D,'cvml','olegstolyar@gmail.com');>> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator? On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right? In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above. I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with? I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls. There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though. On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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See my previous suggestion earlier in this thread On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.
I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls.
There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though.
On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com <javascript:_e(%7B%7D,'cvml','olegstolyar@gmail.com');>> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com <javascript:_e(%7B%7D,'cvml','olegstolyar@gmail.com');>> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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The suggestion about switching to SIP.JS? We are considering this but it will take a us a while to get it done and ready for production. In the meantime I was able to enable debugging on JsSip and it seems that we get this error when receiving re-INVITE from FS: rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set audio receive codecs.. It seems that FS is not including codecs info in re-INVITES. JsSip developers are saying that all invites including re-invites should include all the info. What do you think? On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike@jerris.com> wrote:
See my previous suggestion earlier in this thread
On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.
I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls.
There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though.
On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
> On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: > > Hi guys, > > Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm. > > The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference > > I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light. > > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: > ------------------------------------------------------------------------ > INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 > Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr > Max-Forwards: 70 > From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid >;tag=KQecUSr12rSQp > To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i > Call-ID: g8980rbrbk2t45oj5mru > CSeq: 75703945 INVITE > Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> > User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Session-Expires: 120;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 825 > > v=0 > o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## > s=FreeSWITCH > c=IN IP4 ##.##.##.## > t=0 0 > a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > m=audio 22670 RTP/SAVPF 9 126 106 > a=rtpmap:9 G722/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 > a=rtcp-mux > a=rtcp:22670 IN IP4 ##.##.##.## > a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq > a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 > a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 > a=ice-ufrag:5dS3Fzx1Thrmdy9Z > a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s > a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 > ------------------------------------------------------------------------ > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep >
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I think its incredibly easy to switch to sip.js as its pretty close API to jssip, and looking at the invite you posted, it does have a full sdp. jssip has been mostly abandoned for many months (maybe they came back, but it has serious bitrot either way), its not worth the time to mess with something we know is broken when there is an easy alternative we know works.
On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
The suggestion about switching to SIP.JS? We are considering this but it will take a us a while to get it done and ready for production. In the meantime I was able to enable debugging on JsSip and it seems that we get this error when receiving re-INVITE from FS: rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set audio receive codecs..
It seems that FS is not including codecs info in re-INVITES. JsSip developers are saying that all invites including re-invites should include all the info. What do you think?
On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: See my previous suggestion earlier in this thread
On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar@gmail.com <mailto:olegstolyar@gmail.com>> wrote: I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.
I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls.
There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though.
On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com <>> wrote: But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com <>> wrote: session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com <>> wrote: Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com <>> wrote: I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com <>> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid>;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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Thanks Michael! There are some internal reasons why it's not trivial for us to switch to sip.js but as I said we'll be looking into that as well. On a side note, JsSip did come back - they are actively releasing new versions. Also, I have a question about best practices. If a client originates a call to FS, do you think it's better to have the client be the session refresher or FS? Any reason one way or another? On Thu, May 21, 2015 at 7:50 AM, Michael Jerris <mike@jerris.com> wrote:
I think its incredibly easy to switch to sip.js as its pretty close API to jssip, and looking at the invite you posted, it does have a full sdp. jssip has been mostly abandoned for many months (maybe they came back, but it has serious bitrot either way), its not worth the time to mess with something we know is broken when there is an easy alternative we know works.
On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
The suggestion about switching to SIP.JS? We are considering this but it will take a us a while to get it done and ready for production. In the meantime I was able to enable debugging on JsSip and it seems that we get this error when receiving re-INVITE from FS: rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set audio receive codecs..
It seems that FS is not including codecs info in re-INVITES. JsSip developers are saying that all invites including re-invites should include all the info. What do you think?
On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike@jerris.com> wrote:
See my previous suggestion earlier in this thread
On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.
I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls.
There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though.
On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
> I think the sip.js guys fixed this issue when they forked jssip. > I'd suggest using that. > > > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > > > > Hi guys, > > > > Several weeks ago I started getting an occasional problem where FS > is sending an INVITE to the other side in the middle of a call, the other > side does not respond and FS hangs up the leg. Below is the relevant log. > The user experience is that they keep talking and hearing each other up to > the very end. I have a recording of that call, so can confirm. > > > > The call uses WebRTC and is originated by JsSip from Chrome. Then > the user is put into a conference but I doubt it's relevant in this case > since the INVITE and disconnect are not happening from mod_conference > > > > I suspect it's a re-INVITE but what triggers FS to send it? I > couldn't find anything in the logs that could shed light. > > > > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: > > > ------------------------------------------------------------------------ > > INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob > SIP/2.0 > > Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr > > Max-Forwards: 70 > > From: < > sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid > >;tag=KQecUSr12rSQp > > To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i > > Call-ID: g8980rbrbk2t45oj5mru > > CSeq: 75703945 INVITE > > Contact: < > sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp > > > > User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, path, replaces > > Session-Expires: 120;refresher=uac > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 825 > > > > v=0 > > o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## > > s=FreeSWITCH > > c=IN IP4 ##.##.##.## > > t=0 0 > > a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > > m=audio 22670 RTP/SAVPF 9 126 106 > > a=rtpmap:9 G722/8000 > > a=rtpmap:126 telephone-event/8000 > > a=rtpmap:106 CN/8000 > > a=ptime:20 > > a=fingerprint:sha-256 > E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 > > a=rtcp-mux > > a=rtcp:22670 IN IP4 ##.##.##.## > > a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq > > a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 > > a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > > a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 > > a=ice-ufrag:5dS3Fzx1Thrmdy9Z > > a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s > > a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ > host generation 0 > > > ------------------------------------------------------------------------ > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 > [DEBUG] sofia.c:6627 Channel > sofia/leia_agent/anonymous@anonymous.invalid entering state > [calling][0] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] sofia.c:6627 Channel > sofia/leia_agent/anonymous@anonymous.invalid entering state > [terminating][503] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [NOTICE] sofia.c:7543 Hangup > sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] > [NORMAL_TEMPORARY_FAILURE] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_channel.c:3242 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [KILL] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1396 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: > NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_conference.c:9650 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [UNBRIDGE] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_media.c:7772 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [REFER_EVENT] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_codec.c:246 > sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec > G722:9. > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_session.c:2901 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:535 ( > sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going > to sleep > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:472 ( > sofia/leia_agent/anonymous@anonymous.invalid) Running State Change > CS_HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:735 ( > sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change > ACTIVE -> HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:737 ( > sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_sofia.c:413 Channel > sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: > NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:60 > sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, > cause: NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:737 ( > sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to > sleep > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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Hi Michael, I followed your suggestion and tested the same call using sip.js. You are right sip.js does not disconnect the call after receiving the re-invite. However, it does send back the message below. I agree that it's a more graceful way to handle the problem but the original problem on both JsSip and SIP.JS is still that there is something wrong with the re-INVITE message from FS. By the way a side effect of this is that FS does not send any more re-invites for this session after that first attempt. I'll be happy to run any other tests that could help fix this. For now I will either switch to sip.js or use JsSip's new session timer capability which makes JsSip send regular update messages, so FS does not need to send re-invites at all. ------------------------------------------------------------------------ recv 379 bytes from wss/[69.53.236.236]:17084 at 19:11:35.832686: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WSS ##.##.##.##:5067;branch=z9hG4bK9rtFZNj1vK09D To: <sip:oleg1234@something.com>;tag=c6tjtcraq7 From: <sip:echo-test@something.com>;tag=c29e7F87NvQNp Call-ID: 1v8ll5j5dcqj1d2fimo1 CSeq: 75781683 INVITE Warning: 399 sipjs "Cannot update media description" Supported: outbound User-Agent: SIP.js/0.7.0 Content-Length: 0 On Thu, May 21, 2015 at 7:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael!
There are some internal reasons why it's not trivial for us to switch to sip.js but as I said we'll be looking into that as well. On a side note, JsSip did come back - they are actively releasing new versions.
Also, I have a question about best practices. If a client originates a call to FS, do you think it's better to have the client be the session refresher or FS? Any reason one way or another?
On Thu, May 21, 2015 at 7:50 AM, Michael Jerris <mike@jerris.com> wrote:
I think its incredibly easy to switch to sip.js as its pretty close API to jssip, and looking at the invite you posted, it does have a full sdp. jssip has been mostly abandoned for many months (maybe they came back, but it has serious bitrot either way), its not worth the time to mess with something we know is broken when there is an easy alternative we know works.
On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
The suggestion about switching to SIP.JS? We are considering this but it will take a us a while to get it done and ready for production. In the meantime I was able to enable debugging on JsSip and it seems that we get this error when receiving re-INVITE from FS: rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set audio receive codecs..
It seems that FS is not including codecs info in re-INVITES. JsSip developers are saying that all invites including re-invites should include all the info. What do you think?
On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike@jerris.com> wrote:
See my previous suggestion earlier in this thread
On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all. It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
In most cases when it's sent JsSip answers with a 100 Trying. In this case the call stays on. In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.
I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case. The ACK is there for these calls.
There is a variable nonce-ttl in the sip profile that defaults to 60 seconds. Perhaps it has something to do with that? This profile does not require authentication or registration at all, though.
On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Thanks Michael, I'll see if we can do that! > > So, is the re-INVITE legit and the problem is that JsSip does not > respond to it? Still, I am curious what is triggering the re-INVITE. > > On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> > wrote: > >> I think the sip.js guys fixed this issue when they forked jssip. >> I'd suggest using that. >> >> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> >> wrote: >> > >> > Hi guys, >> > >> > Several weeks ago I started getting an occasional problem where >> FS is sending an INVITE to the other side in the middle of a call, the >> other side does not respond and FS hangs up the leg. Below is the relevant >> log. The user experience is that they keep talking and hearing each other >> up to the very end. I have a recording of that call, so can confirm. >> > >> > The call uses WebRTC and is originated by JsSip from Chrome. >> Then the user is put into a conference but I doubt it's relevant in this >> case since the INVITE and disconnect are not happening from mod_conference >> > >> > I suspect it's a re-INVITE but what triggers FS to send it? I >> couldn't find anything in the logs that could shed light. >> > >> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: >> > >> ------------------------------------------------------------------------ >> > INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob >> SIP/2.0 >> > Via: SIP/2.0/WSS 10.97.158.232:5067 >> ;branch=z9hG4bK7Xm4tjevU45Sr >> > Max-Forwards: 70 >> > From: < >> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid >> >;tag=KQecUSr12rSQp >> > To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i >> > Call-ID: g8980rbrbk2t45oj5mru >> > CSeq: 75703945 INVITE >> > Contact: < >> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp >> > >> > User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit >> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, >> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> > Supported: timer, path, replaces >> > Session-Expires: 120;refresher=uac >> > Min-SE: 120 >> > Content-Type: application/sdp >> > Content-Disposition: session >> > Content-Length: 825 >> > >> > v=0 >> > o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## >> > s=FreeSWITCH >> > c=IN IP4 ##.##.##.## >> > t=0 0 >> > a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW >> > m=audio 22670 RTP/SAVPF 9 126 106 >> > a=rtpmap:9 G722/8000 >> > a=rtpmap:126 telephone-event/8000 >> > a=rtpmap:106 CN/8000 >> > a=ptime:20 >> > a=fingerprint:sha-256 >> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 >> > a=rtcp-mux >> > a=rtcp:22670 IN IP4 ##.##.##.## >> > a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq >> > a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 >> > a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW >> > a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 >> > a=ice-ufrag:5dS3Fzx1Thrmdy9Z >> > a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s >> > a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ >> host generation 0 >> > >> ------------------------------------------------------------------------ >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 >> [DEBUG] switch_core_session.c:1061 Send signal >> sofia/leia_agent/anonymous@anonymous.invalid [BREAK] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 >> [DEBUG] sofia.c:6627 Channel >> sofia/leia_agent/anonymous@anonymous.invalid entering state >> [calling][0] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [DEBUG] switch_core_session.c:1061 Send signal >> sofia/leia_agent/anonymous@anonymous.invalid [BREAK] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [DEBUG] switch_core_session.c:1061 Send signal >> sofia/leia_agent/anonymous@anonymous.invalid [BREAK] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [DEBUG] sofia.c:6627 Channel >> sofia/leia_agent/anonymous@anonymous.invalid entering state >> [terminating][503] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [NOTICE] sofia.c:7543 Hangup >> sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] >> [NORMAL_TEMPORARY_FAILURE] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [DEBUG] switch_channel.c:3242 Send signal >> sofia/leia_agent/anonymous@anonymous.invalid [KILL] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 >> [DEBUG] switch_core_session.c:1396 Send signal >> sofia/leia_agent/anonymous@anonymous.invalid [BREAK] >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: >> NORMAL_TEMPORARY_FAILURE >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] mod_conference.c:9650 >> sofia/leia_agent/anonymous@anonymous.invalid skip receive message >> [UNBRIDGE] (channel is hungup already) >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_media.c:7772 >> sofia/leia_agent/anonymous@anonymous.invalid skip receive message >> [REFER_EVENT] (channel is hungup already) >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_codec.c:246 >> sofia/leia_agent/anonymous@anonymous.invalid Restore previous >> codec G722:9. >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_session.c:2901 >> sofia/leia_agent/anonymous@anonymous.invalid skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:535 ( >> sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going >> to sleep >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:472 ( >> sofia/leia_agent/anonymous@anonymous.invalid) Running State Change >> CS_HANGUP >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:735 ( >> sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change >> ACTIVE -> HANGUP >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:737 ( >> sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] mod_sofia.c:413 Channel >> sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: >> NORMAL_TEMPORARY_FAILURE >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:60 >> sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, >> cause: NORMAL_TEMPORARY_FAILURE >> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 >> [DEBUG] switch_core_state_machine.c:737 ( >> sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going >> to sleep >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >
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The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120 Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040 On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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Thanks Steven! It may be https://freeswitch.org/jira/browse/FS-7040. As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed. INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754 SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid
;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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Just FYI, the JsSip engineers replied to my questions with this: The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass): On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Steven!
It may be https://freeswitch.org/jira/browse/FS-7040.
As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754
SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid
;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
> On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: > > Hi guys, > > Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm. > > The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference > > I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light. > > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: > ------------------------------------------------------------------------ > INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 > Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr > Max-Forwards: 70 > From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid >;tag=KQecUSr12rSQp > To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i > Call-ID: g8980rbrbk2t45oj5mru > CSeq: 75703945 INVITE > Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@ ##.##.###.###:5080;transport=udp> > User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Session-Expires: 120;refresher=uac > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 825 > > v=0 > o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## > s=FreeSWITCH > c=IN IP4 ##.##.##.## > t=0 0 > a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > m=audio 22670 RTP/SAVPF 9 126 106 > a=rtpmap:9 G722/8000 > a=rtpmap:126 telephone-event/8000 > a=rtpmap:106 CN/8000 > a=ptime:20 > a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 > a=rtcp-mux > a=rtcp:22670 IN IP4 ##.##.##.## > a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq > a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 > a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 > a=ice-ufrag:5dS3Fzx1Thrmdy9Z > a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s > a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 > ------------------------------------------------------------------------ > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep >
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Please file a jira for us on that. Even better with a pull request to fix it.
On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Just FYI, the JsSip engineers replied to my questions with this:
The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass):
On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com <mailto:olegstolyar@gmail.com>> wrote: Thanks Steven!
It may be https://freeswitch.org/jira/browse/FS-7040 <https://freeswitch.org/jira/browse/FS-7040>.
As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754
SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid>;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com <mailto:steveayre@gmail.com>> wrote: The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040 <https://freeswitch.org/jira/browse/FS-7040>
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com <mailto:olegstolyar@gmail.com>> wrote: But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com <mailto:olegstolyar@gmail.com>> wrote: Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com <>> wrote: I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com <>> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid>;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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Jira: https://freeswitch.org/jira/browse/FS-7582 It turns out that FS actually does not send a=setup at all in reINVITES. I am not comfortable enough with the code to add an SDP attribute. Also this is now a low priority for me since there is a simple workaround of disabling session timers. On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike@jerris.com> wrote:
Please file a jira for us on that. Even better with a pull request to fix it.
On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Just FYI, the JsSip engineers replied to my questions with this:
The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass):
On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Steven!
It may be https://freeswitch.org/jira/browse/FS-7040.
As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754
SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid
;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
> I think the sip.js guys fixed this issue when they forked jssip. > I'd suggest using that. > > > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > > > > Hi guys, > > > > Several weeks ago I started getting an occasional problem where FS > is sending an INVITE to the other side in the middle of a call, the other > side does not respond and FS hangs up the leg. Below is the relevant log. > The user experience is that they keep talking and hearing each other up to > the very end. I have a recording of that call, so can confirm. > > > > The call uses WebRTC and is originated by JsSip from Chrome. Then > the user is put into a conference but I doubt it's relevant in this case > since the INVITE and disconnect are not happening from mod_conference > > > > I suspect it's a re-INVITE but what triggers FS to send it? I > couldn't find anything in the logs that could shed light. > > > > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: > > > ------------------------------------------------------------------------ > > INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob > SIP/2.0 > > Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr > > Max-Forwards: 70 > > From: < > sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid > >;tag=KQecUSr12rSQp > > To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i > > Call-ID: g8980rbrbk2t45oj5mru > > CSeq: 75703945 INVITE > > Contact: < > sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp > > > > User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, > UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Supported: timer, path, replaces > > Session-Expires: 120;refresher=uac > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 825 > > > > v=0 > > o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## > > s=FreeSWITCH > > c=IN IP4 ##.##.##.## > > t=0 0 > > a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > > m=audio 22670 RTP/SAVPF 9 126 106 > > a=rtpmap:9 G722/8000 > > a=rtpmap:126 telephone-event/8000 > > a=rtpmap:106 CN/8000 > > a=ptime:20 > > a=fingerprint:sha-256 > E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 > > a=rtcp-mux > > a=rtcp:22670 IN IP4 ##.##.##.## > > a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq > > a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 > > a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW > > a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 > > a=ice-ufrag:5dS3Fzx1Thrmdy9Z > > a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s > > a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ > host generation 0 > > > ------------------------------------------------------------------------ > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 > [DEBUG] sofia.c:6627 Channel > sofia/leia_agent/anonymous@anonymous.invalid entering state > [calling][0] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1061 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] sofia.c:6627 Channel > sofia/leia_agent/anonymous@anonymous.invalid entering state > [terminating][503] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [NOTICE] sofia.c:7543 Hangup > sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] > [NORMAL_TEMPORARY_FAILURE] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_channel.c:3242 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [KILL] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 > [DEBUG] switch_core_session.c:1396 Send signal > sofia/leia_agent/anonymous@anonymous.invalid [BREAK] > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: > NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_conference.c:9650 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [UNBRIDGE] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_media.c:7772 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [REFER_EVENT] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_codec.c:246 > sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec > G722:9. > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_session.c:2901 > sofia/leia_agent/anonymous@anonymous.invalid skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:535 ( > sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going > to sleep > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:472 ( > sofia/leia_agent/anonymous@anonymous.invalid) Running State Change > CS_HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:735 ( > sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change > ACTIVE -> HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:737 ( > sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] mod_sofia.c:413 Channel > sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: > NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:60 > sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, > cause: NORMAL_TEMPORARY_FAILURE > > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 > [DEBUG] switch_core_state_machine.c:737 ( > sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to > sleep > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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https://freeswitch.org/jira/browse/FS-7432 There is also this. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Oleg Stolyar Sent: Wednesday, May 27, 2015 12:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call Jira: https://freeswitch.org/jira/browse/FS-7582 It turns out that FS actually does not send a=setup at all in reINVITES. I am not comfortable enough with the code to add an SDP attribute. Also this is now a low priority for me since there is a simple workaround of disabling session timers. On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: Please file a jira for us on that. Even better with a pull request to fix it. On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Just FYI, the JsSip engineers replied to my questions with this: The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass): On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Thanks Steven! It may be https://freeswitch.org/jira/browse/FS-7040. As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed. INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754 SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid>;party=calling;privacy=off;screen=no On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com<mailto:steveayre@gmail.com>> wrote: The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120 Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040 On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator? On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: session timer On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Thanks Michael, I'll see if we can do that! So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE. On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid>;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State HANGUP going to sleep
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You are right Sharath! I resolved mine as a duplicate. On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar <Sharath.Kumar@mezocliq.com> wrote:
https://freeswitch.org/jira/browse/FS-7432
There is also this.
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Oleg Stolyar *Sent:* Wednesday, May 27, 2015 12:14 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a call
Jira: https://freeswitch.org/jira/browse/FS-7582
It turns out that FS actually does not send a=setup at all in reINVITES. I am not comfortable enough with the code to add an SDP attribute. Also this is now a low priority for me since there is a simple workaround of disabling session timers.
On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike@jerris.com> wrote:
Please file a jira for us on that. Even better with a pull request to fix it.
On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Just FYI, the JsSip engineers replied to my questions with this:
The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass):
On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Steven!
It may be https://freeswitch.org/jira/browse/FS-7040.
As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
INVITE sip:echo-test@anonymous.invalid SIP/2.0
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
Max-Forwards: 69
To: <sip:echo-test@anonymous.invalid>
From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2754
SIP/2.0 200 OK
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2
To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:echo-test@50.19.255.62:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 809
X-Netflix: yes
Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid
;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes.
Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: < sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: < sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 ( sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 ( sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 ( sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 ( sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 ( sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Thanks Oleg. Disabling session timers is okay to fix it for regular calls albeit without session timers. But there is also the case of hold/retrieve and other mid-call offer/answers, like say adding someone into a conference etc, where this problem arises. This is a complicated problem. I am yet to test the fix. But chrome has some outstanding problems with role changes as well. They don’t fully support the RFC yet. https://code.google.com/p/webrtc/issues/detail?id=2782 "#4 juberti@chromium.org<mailto:juberti@chromium.org> Offers MUST always use actpass, but Chrome does not support role changes. So, if A calls B, A offers actpass, B responds with active. But then if B does a reoffer to A, B has to offer actpass, and A must respond with passive (since A was chosen as passive before by B's choice of active). Naturally, if A reoffers, A offers actpass and B must respond (again) with active." Thank you, Sharath From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Oleg Stolyar Sent: Wednesday, May 27, 2015 4:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call You are right Sharath! I resolved mine as a duplicate. On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar <Sharath.Kumar@mezocliq.com<mailto:Sharath.Kumar@mezocliq.com>> wrote: https://freeswitch.org/jira/browse/FS-7432 There is also this. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Oleg Stolyar Sent: Wednesday, May 27, 2015 12:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call Jira: https://freeswitch.org/jira/browse/FS-7582 It turns out that FS actually does not send a=setup at all in reINVITES. I am not comfortable enough with the code to add an SDP attribute. Also this is now a low priority for me since there is a simple workaround of disabling session timers. On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: Please file a jira for us on that. Even better with a pull request to fix it. On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Just FYI, the JsSip engineers replied to my questions with this: The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass): On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Thanks Steven! It may be https://freeswitch.org/jira/browse/FS-7040. As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed. INVITE sip:echo-test@anonymous.invalid SIP/2.0 Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029 Max-Forwards: 69 To: <sip:echo-test@anonymous.invalid> From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob> Content-Type: application/sdp Session-Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2754 SIP/2.0 200 OK Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200 From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2 To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg Call-ID: r65e48gp171p21rkppcu CSeq: 6621 INVITE Contact: <sip:echo-test@50.19.255.62:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 809 X-Netflix: yes Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid>;party=calling;privacy=off;screen=no On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com<mailto:steveayre@gmail.com>> wrote: The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes. Session-Expires: 120;refresher=uac Min-SE: 120 Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040 On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator? On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: session timer On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote: Thanks Michael, I'll see if we can do that! So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE. On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com<mailto:mike@jerris.com>> wrote: I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com<mailto:olegstolyar@gmail.com>> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933: ------------------------------------------------------------------------ INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid>;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0 ------------------------------------------------------------------------ fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous@anonymous.invalid<mailto:sofia/leia_agent/anonymous@anonymous.invalid>) State HANGUP going to sleep
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Sure Sharath, I only meant that the workaround works for me since I don't need the mid-call offers/answers. On Wed, May 27, 2015 at 1:47 PM, Sharath Kumar <Sharath.Kumar@mezocliq.com> wrote:
Thanks Oleg. Disabling session timers is okay to fix it for regular calls albeit without session timers. But there is also the case of hold/retrieve and other mid-call offer/answers, like say adding someone into a conference etc, where this problem arises. This is a complicated problem. I am yet to test the fix. But chrome has some outstanding problems with role changes as well. They don’t fully support the RFC yet.
https://code.google.com/p/webrtc/issues/detail?id=2782
"#4 juberti@chromium.org Offers MUST always use actpass, but Chrome does not support role changes. So, if A calls B, A offers actpass, B responds with active. But then if B does a reoffer to A, B has to offer actpass, and A must respond with passive (since A was chosen as passive before by B's choice of active). Naturally, if A reoffers, A offers actpass and B must respond (again) with active."
Thank you,
Sharath
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Oleg Stolyar *Sent:* Wednesday, May 27, 2015 4:24 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a call
You are right Sharath! I resolved mine as a duplicate.
On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar < Sharath.Kumar@mezocliq.com> wrote:
https://freeswitch.org/jira/browse/FS-7432
There is also this.
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Oleg Stolyar *Sent:* Wednesday, May 27, 2015 12:14 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a call
Jira: https://freeswitch.org/jira/browse/FS-7582
It turns out that FS actually does not send a=setup at all in reINVITES. I am not comfortable enough with the code to add an SDP attribute. Also this is now a low priority for me since there is a simple workaround of disabling session timers.
On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike@jerris.com> wrote:
Please file a jira for us on that. Even better with a pull request to fix it.
On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Just FYI, the JsSip engineers replied to my questions with this:
The error happens because the incoming reINVITE has a=setup:active in the SDP which is a bug in FreeSwitch (the RFC clearly states that the SDP offer MUST have a=setup:actpass):
On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Steven!
It may be https://freeswitch.org/jira/browse/FS-7040.
As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS. I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small". I even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
INVITE sip:echo-test@anonymous.invalid SIP/2.0
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
Max-Forwards: 69
To: <sip:echo-test@anonymous.invalid>
From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:mqthb9dj@pfcnm9rjv5en.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2754
SIP/2.0 200 OK
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
From: "b3N0b2x5YXI" <sip:anonymous@anonymous.invalid>;tag=h1059gcvb2
To: <sip:echo-test@anonymous.invalid>;tag=SgjX0X4arHUFg
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:echo-test@50.19.255.62:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 809
X-Netflix: yes
Remote-Party-ID: "echo-test" <sip:echo-test@anonymous.invalid
;party=calling;privacy=off;screen=no
On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre@gmail.com> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes.
Session-Expires: 120;refresher=uac Min-SE: 120
Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040
On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar@gmail.com> wrote:
But isn't that based on the session-timeout param which defaults to 30 min? My re-invites occur much sooner than 30 min into a call. Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike@jerris.com> wrote:
session timer
On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael, I'll see if we can do that!
So, is the re-INVITE legit and the problem is that JsSip does not respond to it? Still, I am curious what is triggering the re-INVITE.
On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike@jerris.com> wrote:
I think the sip.js guys fixed this issue when they forked jssip. I'd suggest using that.
On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg. Below is the relevant log. The user experience is that they keep talking and hearing each other up to the very end. I have a recording of that call, so can confirm.
The call uses WebRTC and is originated by JsSip from Chrome. Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
I suspect it's a re-INVITE but what triggers FS to send it? I couldn't find anything in the logs that could shed light.
send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
------------------------------------------------------------------------
INVITE sip:dj012n41@u40rf5qikah5.invalid;transport=ws;ob SIP/2.0 Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr Max-Forwards: 70 From: < sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@anonymous.invalid ;tag=KQecUSr12rSQp To: "user1" <sip:anonymous@anonymous.invalid>;tag=v1rlqab64i Call-ID: g8980rbrbk2t45oj5mru CSeq: 75703945 INVITE Contact: < sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 825
v=0 o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.## s=FreeSWITCH c=IN IP4 ##.##.##.## t=0 0 a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW m=audio 22670 RTP/SAVPF 9 126 106 a=rtpmap:9 G722/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69 a=rtcp-mux a=rtcp:22670 IN IP4 ##.##.##.## a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0 a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0 a=ice-ufrag:5dS3Fzx1Thrmdy9Z a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [calling][0] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous@anonymous.invalid entering state [terminating][503] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous@anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous@anonymous.invalid [KILL] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous@anonymous.invalid [BREAK] fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous@anonymous.invalid Restore previous codec G722:9. fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous@anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 ( sofia/leia_agent/anonymous@anonymous.invalid) State EXECUTE going to sleep fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 ( sofia/leia_agent/anonymous@anonymous.invalid) Running State Change CS_HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 ( sofia/leia_agent/anonymous@anonymous.invalid) Callstate Change ACTIVE -> HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 ( sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous@anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous@anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 ( sofia/leia_agent/anonymous@anonymous.invalid) State HANGUP going to sleep
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Hi All, I have a question. Now I record video using mod_av (or mod_vlс). Classical record_session. This works fine. But a heavy load on the server CPU. There are some possibilities to control the quality of the recorded video? For example bit rate, video resolution, etc. Or how i can optimize the performance of friswitch to recording so do not load the CPU? My dialplan for tests <extension name="test5"> <condition field="destination_number" expression="^(6084)$"> <action application="export" data="nolocal:absolute_codec_string=OPUS,VP8"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="export" data="RECORD_TOGGLE_ON_REPEAT=true"/> <action application="export" data="record_concat_video=true"/> <action application="answer"/> <!-- Shouldn't bind the same file name to both legs, if both press *2 it clobbers the file --> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] --> <action application="bind_meta_app" data="2 a i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_a.mp4"/> <action application="bind_meta_app" data="2 b i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_b.mp4"/> <action application="bridge" data="user/1003${sip_profile}"/> </condition> </extension> In vars.xml i try to change <!-- Video Settings --> <!-- Setting the max bandwdith --> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=512kb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=512kb"/> but it does not help Any ideas? -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382
What bitrate of recorded video? Or what length and its weight(megabytes) ? 2015-08-26 19:30 GMT+05:00 Denis Jakovlev <yadenis@seznam.cz>:
Hi All,
I have a question. Now I record video using mod_av (or mod_vlс). Classical record_session. This works fine. But a heavy load on the server CPU. There are some possibilities to control the quality of the recorded video? For example bit rate, video resolution, etc. Or how i can optimize the performance of friswitch to recording so do not load the CPU?
My dialplan for tests
<extension name="test5"> <condition field="destination_number" expression="^(6084)$"> <action application="export" data="nolocal:absolute_codec_string=OPUS,VP8"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="export" data="RECORD_TOGGLE_ON_REPEAT=true"/> <action application="export" data="record_concat_video=true"/> <action application="answer"/> <!-- Shouldn't bind the same file name to both legs, if both press *2 it clobbers the file --> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] --> <action application="bind_meta_app" data="2 a i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_a.mp4"/> <action application="bind_meta_app" data="2 b i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_b.mp4"/> <action application="bridge" data="user/1003${sip_profile}"/> </condition> </extension>
In vars.xml i try to change <!-- Video Settings --> <!-- Setting the max bandwdith --> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=512kb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=512kb"/>
but it does not help
Any ideas?
*-- S pozdravem, Ing.Denis Jakovlev mob.tel <http://mob.tel>. 775-415-382 *
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-- Arthur
Dobrý den, Data rate 1311kbps, Total bitrate 1442kbps, Frame Rate 28 frames/second Frame width = 1280 Frame height = 480 -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 čtvrtek 27. srpna 2015, 13:26:26, napsal jste: What bitrate of recorded video? Or what length and its weight(megabytes) ? 2015-08-26 19:30 GMT+05:00 Denis Jakovlev <yadenis@seznam.cz>: Hi All, I have a question. Now I record video using mod_av (or mod_vlс). Classical record_session. This works fine. But a heavy load on the server CPU. There are some possibilities to control the quality of the recorded video? For example bit rate, video resolution, etc. Or how i can optimize the performance of friswitch to recording so do not load the CPU? My dialplan for tests <extension name="test5"> <condition field="destination_number" expression="^(6084)$"> <action application="export" data="nolocal:absolute_codec_string=OPUS,VP8"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="RECORD_STEREO=true"/> <action application="export" data="RECORD_TOGGLE_ON_REPEAT=true"/> <action application="export" data="record_concat_video=true"/> <action application="answer"/> <!-- Shouldn't bind the same file name to both legs, if both press *2 it clobbers the file --> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] --> <action application="bind_meta_app" data="2 a i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_a.mp4"/> <action application="bind_meta_app" data="2 b i record_session::/usr/local/freeswitch/recordings/${strftime(%Y%m%d_%H%M%S)}_${destination_number}_b.mp4"/> <action application="bridge" data="user/1003${sip_profile}"/> </condition> </extension> In vars.xml i try to change <!-- Video Settings --> <!-- Setting the max bandwdith --> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=512kb"/> <X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=512kb"/> but it does not help Any ideas? -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Arthur
participants (7)
-
Artur Mega
-
Denis Jakovlev
-
Joshua Gigg
-
Michael Jerris
-
Oleg Stolyar
-
Sharath Kumar
-
Steven Ayre