[Freeswitch-users] Error in codec negotiation: What are all the places that specify accepted codecs for outgoing calls?

Michael Jerris mike at jerris.com
Tue Jun 10 19:27:22 MSD 2014 

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 at 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 at 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 at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 at 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 at 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 at 54.203.245.50>;tag=9Z1HtD145Qv3K
>>    To: <sip:0011104555592211972 at 108.59.2.133>
>>    Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0
>>    CSeq: 60838805 INVITE
>>    Contact: <sip:gw+voxbeam_outbound at 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 at 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 at 54.203.245.50>;tag=9Z1HtD145Qv3K
>>    To: <sip:0011104555592211972 at 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 at 54.203.245.50>;tag=9Z1HtD145Qv3K
>>    To: <sip:0011104555592211972 at 108.59.2.133>;tag=9312047651043624688
>>    Call-ID: d7aeb7a1-6ac0-1232-718c-22000ad937f0
>>    CSeq: 60838805 INVITE
>>    Content-Type: application/sdp
>>    Contact: <sip:callee at 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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