Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls?
Given that our provider does this 40% of the time in multiple countries, is there anything we can do to get this working with Freeswitch? It seems logical for this to work: we offered a range of codecs, the other end chose one, then for another part of the call chose another. Both codecs are supported by our end, so why should we reject the second just because another was used first? According to the console, Freeswitch is doing a codec renegotiation process... It's just not accepting anything other than 1 codec at this point. +Mark From: Michael Jerris <mike@jerris.com> Subject: Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls? Date: 10 June 2014 12:42:31 GMT To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Hmm, not sure thats true. You are not supposed to change codecs before the dialog is complete like this, its a violation of sdp o/a. On Jun 10, 2014, at 4:50 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Hi Mark,
Thanks for providing the full log.
While goofy this is perfectly valid behavior. Because your instance of Freeswitch offers pcma and pcmu it's ok for the other side to switch between the two in 183.
A few things to try:
- Limit your outbound offers to pcma or pcmu only (one codec).
- The remote side appears to support 100rel (PRACK). Try enabling it in Freeswitch. While a long shot it's possible that PRACK'ing the initial received 183 might change the codec swap behavior of the remote end.
- Try Freeswitch 1.4 or master. There's been some significant changes in media/SDP behavior, particularly when the remote end does technically standards compliant but goofy things with multiple codec offers.
If none of these works you're only option may be to have the carrier take some action.
On Monday, June 9, 2014, Mark Boots <mark@votomobile.org> wrote: Hi Kristian,
I've narrowed this INCOMPATIBLE_DESTINATION error down to a specific case:
1 - Freeswitch sends the outgoing call, and offers PCMU/PCMA/GSM 2 - Other end accepts only PCMU 3 - At some point after Pre-answer, the other end issues a 183 Session Progress, changing to only accept PCMA. 4 - Freeswitch attempts to select new codecs but only looks at PCMU, and cancels the call with INCOMPATIBLE_DESTINATION.
Here is the console log and sip trace... Hope it helps!
+Mark
2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_request_uuid]=[004fd8a2-f01e-11e3-844d-22000ad937f0] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_answer_url]=[http://10.8.0.1/deliverylogs/answer/752471] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_ring_url]=[http://127.0.0.1/callqueue/ring.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/callqueue/hangup.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [origination_caller_id_number]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_from]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_to]=[0011104555592211972] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_app]=[true] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [originate_timeout]=[60] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [ignore_early_media]=[true] 2014-06-09 21:35:39.615207 [NOTICE] switch_channel.c:1049 New Channel sofia/external/0011104555592211972 [005052e6-f01e-11e3-bbae-cbf0f4eac998] 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:5237 (sofia/external/0011104555592211972) State Change CS_NEW -> CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:87 sofia/external/0011104555592211972 SOFIA INIT 2014-06-09 21:35:39.615207 [DEBUG] sofia_glue.c:2746 sofia/external/0011104555592211972 sending invite version: 1.2.24 git 7b74ee3 2014-06-03 00:26:24Z 32bit Local SDP: v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:127 (sofia/external/0011104555592211972) State Change CS_INIT -> CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT going to sleep send 1071 bytes to udp/[108.59.2.133]:5060 at 21:35:39.620673: ------------------------------------------------------------------------ INVITE sip:0011104555592211972@108.59.2.133 SIP/2.0 Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB Max-Forwards: 70 From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Contact: <sip:gw+voxbeam_outbound@54.203.245.50:5080;transport=udp;gw=voxbeam_outbound> User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140603T002624Z~7b74ee3955~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:12026004299@54.203.245.50>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [calling][0] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:150 sofia/external/0011104555592211972 SOFIA ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0011104555592211972) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING going to sleep 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA going to sleep recv 367 bytes from udp/[108.59.2.133]:5060 at 21:35:39.725204: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 54.203.245.50:5080;received=54.203.245.50;rport=5080;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Server: OpenSIPS (1.8.2-notls (x86_64/linux)) Content-Length: 0
------------------------------------------------------------------------ recv 657 bytes from udp/[108.59.2.133]:5060 at 21:35:41.191275: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133>;tag=9312047651043624688 Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Content-Type: application/sdp Contact: <sip:callee@108.59.2.133;did=795.65454242> Supported: timer,100rel Content-Length: 226
v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [proceeding][183] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5858 Remote SDP: v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3192 Set Codec sofia/external/0011104555592211972 PCMU/8000 20 ms 160 samples 64000 bits 2014-06-09 21:35:41.175213 [DEBUG] switch_core_codec.c:111 sofia/external/0011104555592211972 Original read codec set to PCMU:0 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5444 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3451 AUDIO RTP [sofia/external/0011104555592211972] 10.217.55.240 port 24598 -> 80.84.30.41 port 42108 codec: 0 ms: 20 2014-06-09 21:35:41.175213 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3718 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3724 Set 2833 dtmf receive payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3751 sofia/external/0011104555592211972 Set rtp dtmf delay to 40 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3757 Set comfort noise payload to 13 2014-06-09 21:35:41.175213 [NOTICE] sofia_glue.c:4362 Pre-Answer sofia/external/0011104555592211972!
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I did miss that this was proxy forked. That is of course legal, whats not ok is to change codecs on the same dialog before its fully set up. On Jun 10, 2014, at 3:12 PM, Mark Boots <mark@votomobile.org> wrote:
Given that our provider does this 40% of the time in multiple countries, is there anything we can do to get this working with Freeswitch?
It seems logical for this to work: we offered a range of codecs, the other end chose one, then for another part of the call chose another. Both codecs are supported by our end, so why should we reject the second just because another was used first?
According to the console, Freeswitch is doing a codec renegotiation process... It's just not accepting anything other than 1 codec at this point.
+Mark
From: Michael Jerris <mike@jerris.com> Subject: Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls? Date: 10 June 2014 12:42:31 GMT To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Hmm, not sure thats true. You are not supposed to change codecs before the dialog is complete like this, its a violation of sdp o/a.
On Jun 10, 2014, at 4:50 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Hi Mark,
Thanks for providing the full log.
While goofy this is perfectly valid behavior. Because your instance of Freeswitch offers pcma and pcmu it's ok for the other side to switch between the two in 183.
A few things to try:
- Limit your outbound offers to pcma or pcmu only (one codec).
- The remote side appears to support 100rel (PRACK). Try enabling it in Freeswitch. While a long shot it's possible that PRACK'ing the initial received 183 might change the codec swap behavior of the remote end.
- Try Freeswitch 1.4 or master. There's been some significant changes in media/SDP behavior, particularly when the remote end does technically standards compliant but goofy things with multiple codec offers.
If none of these works you're only option may be to have the carrier take some action.
On Monday, June 9, 2014, Mark Boots <mark@votomobile.org> wrote: Hi Kristian,
I've narrowed this INCOMPATIBLE_DESTINATION error down to a specific case:
1 - Freeswitch sends the outgoing call, and offers PCMU/PCMA/GSM 2 - Other end accepts only PCMU 3 - At some point after Pre-answer, the other end issues a 183 Session Progress, changing to only accept PCMA. 4 - Freeswitch attempts to select new codecs but only looks at PCMU, and cancels the call with INCOMPATIBLE_DESTINATION.
Here is the console log and sip trace... Hope it helps!
+Mark
2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_request_uuid]=[004fd8a2-f01e-11e3-844d-22000ad937f0] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_answer_url]=[http://10.8.0.1/deliverylogs/answer/752471] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_ring_url]=[http://127.0.0.1/callqueue/ring.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/callqueue/hangup.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [origination_caller_id_number]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_from]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_to]=[0011104555592211972] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_app]=[true] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [originate_timeout]=[60] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [ignore_early_media]=[true] 2014-06-09 21:35:39.615207 [NOTICE] switch_channel.c:1049 New Channel sofia/external/0011104555592211972 [005052e6-f01e-11e3-bbae-cbf0f4eac998] 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:5237 (sofia/external/0011104555592211972) State Change CS_NEW -> CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:87 sofia/external/0011104555592211972 SOFIA INIT 2014-06-09 21:35:39.615207 [DEBUG] sofia_glue.c:2746 sofia/external/0011104555592211972 sending invite version: 1.2.24 git 7b74ee3 2014-06-03 00:26:24Z 32bit Local SDP: v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:127 (sofia/external/0011104555592211972) State Change CS_INIT -> CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT going to sleep send 1071 bytes to udp/[108.59.2.133]:5060 at 21:35:39.620673: ------------------------------------------------------------------------ INVITE sip:0011104555592211972@108.59.2.133 SIP/2.0 Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB Max-Forwards: 70 From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Contact: <sip:gw+voxbeam_outbound@54.203.245.50:5080;transport=udp;gw=voxbeam_outbound> User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140603T002624Z~7b74ee3955~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:12026004299@54.203.245.50>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [calling][0] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:150 sofia/external/0011104555592211972 SOFIA ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0011104555592211972) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING going to sleep 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA going to sleep recv 367 bytes from udp/[108.59.2.133]:5060 at 21:35:39.725204: ------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 54.203.245.50:5080;received=54.203.245.50;rport=5080;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Server: OpenSIPS (1.8.2-notls (x86_64/linux)) Content-Length: 0
------------------------------------------------------------------------ recv 657 bytes from udp/[108.59.2.133]:5060 at 21:35:41.191275: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133>;tag=9312047651043624688 Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Content-Type: application/sdp Contact: <sip:callee@108.59.2.133;did=795.65454242> Supported: timer,100rel Content-Length: 226
v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [proceeding][183] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5858 Remote SDP: v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3192 Set Codec sofia/external/0011104555592211972 PCMU/8000 20 ms 160 samples 64000 bits 2014-06-09 21:35:41.175213 [DEBUG] switch_core_codec.c:111 sofia/external/0011104555592211972 Original read codec set to PCMU:0 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5444 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3451 AUDIO RTP [sofia/external/0011104555592211972] 10.217.55.240 port 24598 -> 80.84.30.41 port 42108 codec: 0 ms: 20 2014-06-09 21:35:41.175213 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3718 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3724 Set 2833 dtmf receive payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3751 sofia/external/0011104555592211972 Set rtp dtmf delay to 40 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3757 Set comfort noise payload to 13 2014-06-09 21:35:41.175213 [NOTICE] sofia_glue.c:4362 Pre-Answer sofia/external/0011104555592211972!
_______________________________________________ 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___________________________________________________... Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Mark, Have you tried any of my original suggestions? You have a good point on the behavior of FreeSWITCH for codec negotiation in this specific scenario and although I can't point to a commit I seem to recall that 1.4/master tweaked this behavior significantly; it may "just work" for you. 100rel might be a good option too but that could come with some other unintended effects. On Tue, Jun 10, 2014 at 11:12 AM, Mark Boots <mark@votomobile.org> wrote:
Given that our provider does this 40% of the time in multiple countries, is there anything we can do to get this working with Freeswitch?
It seems logical for this to work: we offered a range of codecs, the other end chose one, then for another part of the call chose another. Both codecs are supported by our end, so why should we reject the second just because another was used first?
According to the console, Freeswitch is doing a codec renegotiation process... It's just not accepting anything other than 1 codec at this point.
+Mark
From: Michael Jerris <mike@jerris.com> Subject: Re: [Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls? Date: 10 June 2014 12:42:31 GMT To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Reply-To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Hmm, not sure thats true. You are not supposed to change codecs before the dialog is complete like this, its a violation of sdp o/a.
On Jun 10, 2014, at 4:50 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Hi Mark,
Thanks for providing the full log.
While goofy this is perfectly valid behavior. Because your instance of Freeswitch offers pcma and pcmu it's ok for the other side to switch between the two in 183.
A few things to try:
- Limit your outbound offers to pcma or pcmu only (one codec).
- The remote side appears to support 100rel (PRACK). Try enabling it in Freeswitch. While a long shot it's possible that PRACK'ing the initial received 183 might change the codec swap behavior of the remote end.
- Try Freeswitch 1.4 or master. There's been some significant changes in media/SDP behavior, particularly when the remote end does technically standards compliant but goofy things with multiple codec offers.
If none of these works you're only option may be to have the carrier take some action.
On Monday, June 9, 2014, Mark Boots <mark@votomobile.org> wrote:
Hi Kristian,
I've narrowed this INCOMPATIBLE_DESTINATION error down to a specific case:
1 - Freeswitch sends the outgoing call, and offers PCMU/PCMA/GSM 2 - Other end accepts only PCMU 3 - At some point after Pre-answer, the other end issues a 183 Session Progress, changing to only accept PCMA. 4 - Freeswitch attempts to select new codecs but only looks at PCMU, and cancels the call with INCOMPATIBLE_DESTINATION.
Here is the console log and sip trace... Hope it helps!
+Mark
2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:2069 Parsing global variables 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_request_uuid]=[004fd8a2-f01e-11e3-844d-22000ad937f0] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_answer_url]=[http://10.8.0.1/deliverylogs/answer/752471] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_ring_url]=[http://127.0.0.1/callqueue/ring.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/callqueue/hangup.php] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [origination_caller_id_number]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_from]=[12026004299] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_to]=[0011104555592211972] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [plivo_app]=[true] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [originate_timeout]=[60] 2014-06-09 21:35:39.615207 [DEBUG] switch_event.c:1661 Parsing variable [ignore_early_media]=[true] 2014-06-09 21:35:39.615207 [NOTICE] switch_channel.c:1049 New Channel sofia/external/0011104555592211972 [005052e6-f01e-11e3-bbae-cbf0f4eac998] 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:5237 (sofia/external/0011104555592211972) State Change CS_NEW -> CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_INIT 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:87 sofia/external/0011104555592211972 SOFIA INIT 2014-06-09 21:35:39.615207 [DEBUG] sofia_glue.c:2746 sofia/external/0011104555592211972 sending invite version: 1.2.24 git 7b74ee3 2014-06-03 00:26:24Z 32bit Local SDP: v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:127 (sofia/external/0011104555592211972) State Change CS_INIT -> CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:458 (sofia/external/0011104555592211972) State INIT going to sleep send 1071 bytes to udp/[108.59.2.133]:5060 at 21:35:39.620673:
------------------------------------------------------------------------ INVITE sip:0011104555592211972@108.59.2.133 SIP/2.0 Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB Max-Forwards: 70 From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Contact: <sip:gw+voxbeam_outbound@54.203.245.50:5080;transport=udp;gw=voxbeam_outbound> User-Agent: FreeSWITCH-mod_sofia/1.2.24+git~20140603T002624Z~7b74ee3955~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 207 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:12026004299@54.203.245.50>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1402325141 1402325142 IN IP4 54.203.245.50 s=FreeSWITCH c=IN IP4 54.203.245.50 t=0 0 m=audio 24598 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
------------------------------------------------------------------------ 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_ROUTING 2014-06-09 21:35:39.615207 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [calling][0] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING 2014-06-09 21:35:39.615207 [DEBUG] mod_sofia.c:150 sofia/external/0011104555592211972 SOFIA ROUTING 2014-06-09 21:35:39.615207 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0011104555592211972) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_session.c:1351 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:474 (sofia/external/0011104555592211972) State ROUTING going to sleep 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:418 (sofia/external/0011104555592211972) Running State Change CS_CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA 2014-06-09 21:35:39.615207 [DEBUG] switch_core_state_machine.c:493 (sofia/external/0011104555592211972) State CONSUME_MEDIA going to sleep recv 367 bytes from udp/[108.59.2.133]:5060 at 21:35:39.725204:
------------------------------------------------------------------------ SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 54.203.245.50:5080;received=54.203.245.50;rport=5080;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133> Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Server: OpenSIPS (1.8.2-notls (x86_64/linux)) Content-Length: 0
------------------------------------------------------------------------ recv 657 bytes from udp/[108.59.2.133]:5060 at 21:35:41.191275:
------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 54.203.245.50:5080;rport;branch=z9hG4bKUNeNtZcZp5ZmB From: "" <sip:12026004299@54.203.245.50>;tag=9Z1HtD145Qv3K To: <sip:0011104555592211972@108.59.2.133>;tag=9312047651043624688 Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0 CSeq: 60838805 INVITE Content-Type: application/sdp Contact: <sip:callee@108.59.2.133;did=795.65454242> Supported: timer,100rel Content-Length: 226
v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
------------------------------------------------------------------------ 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] switch_core_session.c:1016 Send signal sofia/external/0011104555592211972 [BREAK] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5845 Channel sofia/external/0011104555592211972 entering state [proceeding][183] 2014-06-09 21:35:41.175213 [DEBUG] sofia.c:5858 Remote SDP: v=0 o=RISNEXT02 869 1999 IN IP4 80.84.30.41 s=sip call c=IN IP4 80.84.30.41 t=0 0 m=audio 42108 RTP/AVP 0 13 101 a=rtpmap:0 PCMU/8000 a=rtpmap:13 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3192 Set Codec sofia/external/0011104555592211972 PCMU/8000 20 ms 160 samples 64000 bits 2014-06-09 21:35:41.175213 [DEBUG] switch_core_codec.c:111 sofia/external/0011104555592211972 Original read codec set to PCMU:0 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:5444 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3451 AUDIO RTP [sofia/external/0011104555592211972] 10.217.55.240 port 24598 -> 80.84.30.41 port 42108 codec: 0 ms: 20 2014-06-09 21:35:41.175213 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3718 Set 2833 dtmf send payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3724 Set 2833 dtmf receive payload to 101 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3751 sofia/external/0011104555592211972 Set rtp dtmf delay to 40 2014-06-09 21:35:41.175213 [DEBUG] sofia_glue.c:3757 Set comfort noise payload to 13 2014-06-09 21:35:41.175213 [NOTICE] sofia_glue.c:4362 Pre-Answer sofia/external/0011104555592211972!
_______________________________________________ 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
-- Kristian Kielhofner
participants (3)
-
Kristian Kielhofner
-
Mark Boots
-
Michael Jerris