SDP with RTP SAVPF is rejected on some SIP phones
Hello All, I am working with webrtc and I am having problems with incoming audio. I use : - Kamailio as a SIP and Websocket proxy - Freeswitch as a media server. UA/WebRTC <--> Kamailio <--> Freeswitch For outgoing calls (webrtc->freeswitch) everything works well, but for incoming call(freeswitch->webrtc), my webrtc client was complaining about missing ICE candidates. In my dialplan I added a new line with: <action application="export" data="media_webrtc=true" /> since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF Now, I can receive calls on my webrtc client but it does not work on some sip phones. On sip phones it is rejected with 488 Not Acceptable Here SDP from freeswitch's INVITE looks like : v=0. o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. s=My-SBC. c=IN IP4 MyPublicIP. t=0 0. a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. m=audio 32204 RTP/SAVPF 8 0 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5: 71:21:11:96:AF:0B:DD:6E:79:9A:AC:FF:86:69:89:14:88:70:AA:EE:4A:EB. a=setup:actpass. a=rtcp-mux. a=rtcp:32204 IN IP4 MyPublicIP. a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0. a=ssrc:590458548 mslabel:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. a=ssrc:590458548 label:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmfa0. a=ice-ufrag:akVRJFgs6QHjvUo2. a=ice-pwd:vG0zewmpyrGKmZhm924Jxa1u. a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. a=ptime:20. Maybe I don't understand this correctly, but is that possible to add SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio based on transport towards client (TLS,UDP,TCP,WSS) ? I have tested it on the latest freeswitch 1.4 and 1.6 versions.
nobody uses webrtc and classic calls with freeswitch as a media server and kamailio or opensips as a sip proxy ? I found a solution with rtpengine, but I want to use freeswitch as a media server. 2016-05-06 9:07 GMT+02:00 John Smolka <john.smolka9@gmail.com>:
Hello All,
I am working with webrtc and I am having problems with incoming audio. I use : - Kamailio as a SIP and Websocket proxy - Freeswitch as a media server.
UA/WebRTC <--> Kamailio <--> Freeswitch
For outgoing calls (webrtc->freeswitch) everything works well, but for incoming call(freeswitch->webrtc), my webrtc client was complaining about missing ICE candidates. In my dialplan I added a new line with:
<action application="export" data="media_webrtc=true" />
since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF
Now, I can receive calls on my webrtc client but it does not work on some sip phones. On sip phones it is rejected with 488 Not Acceptable Here
SDP from freeswitch's INVITE looks like :
v=0. o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. s=My-SBC. c=IN IP4 MyPublicIP. t=0 0. a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. m=audio 32204 RTP/SAVPF 8 0 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5: 71:21:11:96:AF:0B:DD:6E:79:9A:AC:FF:86:69:89:14:88:70:AA:EE:4A:EB. a=setup:actpass. a=rtcp-mux. a=rtcp:32204 IN IP4 MyPublicIP. a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0. a=ssrc:590458548 mslabel:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. a=ssrc:590458548 label:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmfa0. a=ice-ufrag:akVRJFgs6QHjvUo2. a=ice-pwd:vG0zewmpyrGKmZhm924Jxa1u. a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. a=ptime:20.
Maybe I don't understand this correctly, but is that possible to add SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio based on transport towards client (TLS,UDP,TCP,WSS) ?
I have tested it on the latest freeswitch 1.4 and 1.6 versions.
Freeswitch can already translate between AVP and SAVPF. Why would you manipulate the sdp? Shouldn't be necessary
On May 12, 2016, at 10:06 AM, John Smolka <john.smolka9@gmail.com> wrote:
nobody uses webrtc and classic calls with freeswitch as a media server and kamailio or opensips as a sip proxy ? I found a solution with rtpengine, but I want to use freeswitch as a media server.
2016-05-06 9:07 GMT+02:00 John Smolka <john.smolka9@gmail.com <mailto:john.smolka9@gmail.com>>: Hello All,
I am working with webrtc and I am having problems with incoming audio. I use : - Kamailio as a SIP and Websocket proxy - Freeswitch as a media server.
UA/WebRTC <--> Kamailio <--> Freeswitch
For outgoing calls (webrtc->freeswitch) everything works well, but for incoming call(freeswitch->webrtc), my webrtc client was complaining about missing ICE candidates. In my dialplan I added a new line with:
<action application="export" data="media_webrtc=true" />
since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF
Now, I can receive calls on my webrtc client but it does not work on some sip phones. On sip phones it is rejected with 488 Not Acceptable Here
SDP from freeswitch's INVITE looks like :
v=0. o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP. s=My-SBC. c=IN IP4 MyPublicIP. t=0 0. a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. m=audio 32204 RTP/SAVPF 8 0 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5:71:21:11:96:AF:0B:DD:6E:79:9A:AC:FF:86:69:89:14:88:70:AA:EE:4A:EB. a=setup:actpass. a=rtcp-mux. a=rtcp:32204 IN IP4 MyPublicIP. a=ssrc:590458548 cname:GLr7dIFWlcfql4sH. a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0. a=ssrc:590458548 mslabel:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf. a=ssrc:590458548 label:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmfa0. a=ice-ufrag:akVRJFgs6QHjvUo2. a=ice-pwd:vG0zewmpyrGKmZhm924Jxa1u. a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0. a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0. a=ptime:20.
Maybe I don't understand this correctly, but is that possible to add SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio based on transport towards client (TLS,UDP,TCP,WSS) ?
I have tested it on the latest freeswitch 1.4 and 1.6 versions.
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participants (2)
-
John Smolka
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Michael Jerris