Hello, I am new to freeswitch and I have just installed the last version of freeswitch. I am doing some tests dialing with sip and when the call is answer I play an file. If I do 10 calls, in 5 or 6 of them I get this error in the console and the audio gets cut like with it was a bandwithd problem. 2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen. Does anyone knows if there is a way to fix this issue ? Thanks Leonardo
Sorry I found this message: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.h... and the <param name="inbound-codec-negotiation" value="scrooge"/> fixed my problem. Now I have another question. What exactly this "scrooge" does ? Is this going to affect the other providers that was working ? Thanks Leonardo On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <telteclistas@gmail.com>wrote:
Hello,
I am new to freeswitch and I have just installed the last version of freeswitch. I am doing some tests dialing with sip and when the call is answer I play an file. If I do 10 calls, in 5 or 6 of them I get this error in the console and the audio gets cut like with it was a bandwithd problem.
2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.
Does anyone knows if there is a way to fix this issue ?
Thanks Leonardo
yes it will. whatever you are talking to is broken. It's up to them to fix the broken timestamps and ptime advertisement. On Tue, Apr 27, 2010 at 11:19 PM, leonardo alves <telteclistas@gmail.com>wrote:
Sorry I found this message:
http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg10032.h... and the <param name="inbound-codec-negotiation" value="scrooge"/> fixed my problem.
Now I have another question. What exactly this "scrooge" does ? Is this going to affect the other providers that was working ? Thanks Leonardo
On Tue, Apr 27, 2010 at 11:27 PM, leonardo alves <telteclistas@gmail.com>wrote:
Hello,
I am new to freeswitch and I have just installed the last version of freeswitch. I am doing some tests dialing with sip and when the call is answer I play an file. If I do 10 calls, in 5 or 6 of them I get this error in the console and the audio gets cut like with it was a bandwithd problem.
2010-04-27 22:20:56.236782 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.
Does anyone knows if there is a way to fix this issue ?
Thanks Leonardo
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Actually, I tougth it was working but it is not. If when I answer the call and stay in silence everything works ok. But if I say something or there is any noise in the room shows in the freeswitch console the [WARNING] mod_sofia.c:999 and the call starts to get choppy. The command that I am running in the freeswitch console is this: bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/nettophone/18197713136 &playback(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-sample_submenu.wav) Reading the debug of the call what I could understand is while the ptime of the call is in 20ms everything works good, but when it changes to 30ms the call starts to get choppy. And it gets changed after the warning. On the file vars.xml I also tried this configuration: <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU@20i,PCMA@20i"/> But after the warning it still changes the ptime to 30 and the call starts to get choppy. Does anyone knows what else I could do ? Thanks for any ideas. Here is the debug of the call: 2010-04-30 21:40:09.656257 [DEBUG] mod_sofia.c:140 sofia/external/18197713136 SOFIA ROUTING 2010-04-30 21:40:09.656257 [DEBUG] switch_ivr_originate.c:66 (sofia/external/18197713136) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2010-04-30 21:40:09.656257 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/18197713136 [BREAK] 2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:341 (sofia/external/18197713136) State ROUTING going to sleep 2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:314 (sofia/external/18197713136) Running State Change CS_CONSUME_MEDIA 2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:360 (sofia/external/18197713136) State CONSUME_MEDIA 2010-04-30 21:40:09.656257 [DEBUG] switch_core_state_machine.c:360 (sofia/external/18197713136) State CONSUME_MEDIA going to sleep send 1085 bytes to udp/[66.33.157.119]:5060 at 02:40:09.657682: ------------------------------------------------------------------------ INVITE sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKBB4KpNNHQUFyS Max-Forwards: 70 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>> Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222468 INVITE Contact: <sip:gw+nettophone@10.1.1.1:5080;transport=udp;gw=nettophone> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-FS-Support: update_display Remote-Party-ID: <sip:0000000000@66.33.157.119<sip%3A0000000000@66.33.157.119>
;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1272653081 1272653082 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 28528 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20
2010-04-30 21:40:09.656257 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [calling][0] recv 411 bytes from udp/[66.33.157.119]:5060 at 02:40:09.703164: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.1:5080 ;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222468 INVITE Contact: <sip:66.33.157.119:5060> Server: Net2Phone Carrier Content-Length: 0
recv 657 bytes from udp/[66.33.157.119]:5060 at 02:40:11.008062: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.1.1.1:5080 ;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222468 INVITE Contact: <sip:66.33.157.119:5060> Server: Net2Phone Carrier Content-Length: 203 Content-Type: application/sdp v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
2010-04-30 21:40:11.008120 [INFO] sofia.c:662 Update Callee ID to "18197713136" <18197713136> 2010-04-30 21:40:11.008120 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [proceeding][183] 2010-04-30 21:40:11.008120 [DEBUG] sofia.c:4183 Remote SDP: v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:3674 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:2372 Set Codec sofia/external/18197713136 PCMU/8000 20 ms 160 samples 2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:3607 Set 2833 dtmf send payload to 101 2010-04-30 21:40:11.008120 [DEBUG] sofia_glue.c:2612 AUDIO RTP [sofia/external/18197713136] 10.1.1.1 port 28528 -> 169.132.188.43 port 22696 codec: 0 ms: 20 2010-04-30 21:40:11.008120 [DEBUG] switch_rtp.c:1346 Starting timer [soft] 160 bytes per 20ms 2010-04-30 21:40:11.012070 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send payload to 101 2010-04-30 21:40:11.012070 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf receive payload to 101 2010-04-30 21:40:11.012070 [NOTICE] sofia_glue.c:3227 Pre-Answer sofia/external/18197713136! recv 681 bytes from udp/[66.33.157.119]:5060 at 02:40:18.227385: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5080 ;branch=z9hG4bKBB4KpNNHQUFyS;received=10.1.1.1;rport=5080 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Allow: ACK,BYE,CANCEL,INVITE,OPTIONS Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222468 INVITE Contact: <sip:66.33.157.119:5060> Server: Net2Phone Carrier Content-Length: 203 Content-Type: application/sdp v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [completing][200] 2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4180 Duplicate SDP v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 send 429 bytes to udp/[66.33.157.119]:5060 at 02:40:18.228604: ------------------------------------------------------------------------ ACK sip:66.33.157.119:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKcmXcrg6mm45gN Max-Forwards: 70 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222468 ACK Contact: <sip:gw+nettophone@10.1.1.1:5080;transport=udp;gw=nettophone> Content-Length: 0
2010-04-30 21:40:18.227023 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [ready][200] 2010-04-30 21:40:18.227023 [NOTICE] sofia.c:4696 Channel [sofia/external/18197713136] has been answered 2010-04-30 21:40:18.228967 [DEBUG] switch_ivr_originate.c:3210 Originate Resulted in Success: [sofia/external/18197713136] 2010-04-30 21:40:18.228967 [DEBUG] mod_commands.c:2870 (sofia/external/18197713136) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2010-04-30 21:40:18.228967 [DEBUG] switch_core_session.c:1022 Send signal sofia/external/18197713136 [BREAK] 2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:314 (sofia/external/18197713136) Running State Change CS_EXECUTE 2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:348 (sofia/external/18197713136) State EXECUTE 2010-04-30 21:40:18.228967 [DEBUG] mod_sofia.c:233 sofia/external/18197713136 SOFIA EXECUTE 2010-04-30 21:40:18.228967 [DEBUG] switch_core_state_machine.c:157 sofia/external/18197713136 Standard EXECUTE EXECUTE sofia/external/18197713136 playback(/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-sample_submenu.wav) 2010-04-30 21:40:18.229973 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-04-30 21:40:18.287128 [DEBUG] switch_rtp.c:2446 Correct ip/port confirmed. 2010-04-30 21:40:18.486993 [WARNING] mod_sofia.c:999 We were told to use ptime 20 but what they meant to say was 30 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-04-30 21:40:18.486993 [DEBUG] sofia_glue.c:2301 Changing Codec from PCMU@20ms to PCMU@30ms 2010-04-30 21:40:18.486993 [DEBUG] switch_rtp.c:1232 RE-Starting timer [soft] 240 bytes per 30000ms 2010-04-30 21:40:18.486993 [DEBUG] sofia_glue.c:2372 Set Codec sofia/external/18197713136 PCMU/8000 30 ms 240 samples send 1081 bytes to udp/[66.33.157.119]:5060 at 02:40:18.487972: ------------------------------------------------------------------------ INVITE sip:66.33.157.119:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKDXp5SBQrHDv3g Max-Forwards: 70 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222469 INVITE Contact: <sip:gw+nettophone@10.1.1.1:5080;transport=udp;gw=nettophone> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-Broken-PTIME: Adv=20;Sent=30 X-FS-Support: update_display Remote-Party-ID: <sip:0000000000@66.33.157.119<sip%3A0000000000@66.33.157.119> ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1272653081 1272653083 IN IP4 10.1.1.1 s=FreeSWITCH c=IN IP4 10.1.1.1 t=0 0 m=audio 28528 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30
2010-04-30 21:40:18.486993 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [calling][0] 2010-04-30 21:40:18.517050 [DEBUG] switch_core_io.c:896 Engaging Write Buffer at 480 bytes to accommodate 320->480 recv 681 bytes from udp/[66.33.157.119]:5060 at 02:40:18.534493: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.1:5080 ;branch=z9hG4bKDXp5SBQrHDv3g;received=10.1.1.1;rport=5080 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Allow: ACK,BYE,CANCEL,INVITE,OPTIONS Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222469 INVITE Contact: <sip:66.33.157.119:5060> Server: Net2Phone Carrier Content-Length: 203 Content-Type: application/sdp v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
send 429 bytes to udp/[66.33.157.119]:5060 at 02:40:18.534747: ------------------------------------------------------------------------ ACK sip:66.33.157.119:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5080;rport;branch=z9hG4bKe6FyU67Uepjpc Max-Forwards: 70 From: "" <sip:cust_USERNAME@66.33.157.119<sip%3Acust_USERNAME@66.33.157.119> ;transport=udp>;tag=BBvj27gt2ZFQB To: <sip:18197713136@66.33.157.119 <sip%3A18197713136@66.33.157.119>
;tag=ccid-713620800-1-574 Call-ID: b36e59ea-cf6d-122d-0a9c-0026b97cdbb8 CSeq: 130222469 ACK Contact: <sip:gw+nettophone@10.1.1.1:5080;transport=udp;gw=nettophone> Content-Length: 0
2010-04-30 21:40:18.534975 [DEBUG] sofia.c:4172 Channel sofia/external/18197713136 entering state [ready][200] 2010-04-30 21:40:18.534975 [DEBUG] sofia.c:4180 Duplicate SDP v=0 o=44952 713620800 713620800 IN IP4 169.132.188.43 s=SIP Call c=IN IP4 169.132.188.43 t=0 0 m=audio 22696 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11
participants (2)
-
Anthony Minessale
-
leonardo alves