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Summary:ASTERISK-25428: Codec negotation fails 'No compatible codecs, not accepting this offer!' in transfer scenario between two servers when using SILK and SPEEX
Reporter:Peter Katzmann (pk16208)Labels:
Date Opened:2015-09-28 03:56:01Date Closed:
Priority:MajorRegression?
Status:Open/NewComponents:Channels/chan_sip/CodecHandling
Versions:11.19.0 13.18.4 Frequency of
Occurrence
Related
Issues:
Environment:ubuntu precise and trustyAttachments:( 0) codec_neg_prob.pcap
( 1) servear_a.log
( 2) servear_b.log
( 3) servera_sip.conf
( 4) servera.log
( 5) servera.sip.conf
( 6) serverb_sip.conf
( 7) serverb.log
( 8) serverb.sip.conf
( 9) vmfail2.pcap
Description:There are two Servers (A and B) connected over wan via silk and speex
There is user 7000 on Server A and user 8000 on Server B.
Voicemail will only be handled from server A.
User 8000 has forwarding to voicemail after 5 Seks.

Now User 7000 calls user 8000, when User 8000 picks up teh phone everything is OK.
If the call is expired it will be forwarded to the voicemail server A, in this case the call fails with no acceptable codec offer.

When i modify the sip conf for server A or Server B to alow ulaw/alaw or g722 voicemail works as expected.

Comments:By: Asterisk Team (asteriskteam) 2015-09-28 03:56:05.658-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Peter Katzmann (pk16208) 2015-09-28 03:57:24.174-0500

Logfiles of failing scenario
By: Rusty Newton (rnewton) 2015-10-01 07:17:40.998-0500

It isn't immediately clear what the issue is. Please attach a new pcap and new logs, as well as the sip configuration.

For troubleshooting purposes the calls in the logs should match the calls in the pcap. Please make sure the log additionally includes the output of "sip set debug on"

Thanks!
By: Peter Katzmann (pk16208) 2015-10-01 08:04:32.752-0500

Log files as requested
By: Peter Katzmann (pk16208) 2015-10-01 08:12:09.356-0500

The call to record fails if the allowed codecs are only speex/silk.
The call to record is ok if ulaw/alaw or g722  is allowed

But it fails only when the call is routed back to origin server (in this case from 7000/ServerA over 8000/ServerB to Record/ServerA)


When we change the scenario for eg. to  
7000/ServerA over 8000/ServerB to Record/ServerB
or
7000/ServerA over 7000/ServerB to Record/Server C
it works.
By: Rusty Newton (rnewton) 2015-10-01 14:57:06.530-0500

Thanks! We'll take a deeper look at the new debug when able.
By: Rusty Newton (rnewton) 2015-10-16 17:48:37.744-0500

Yeah I see where the INVITE comes in to asterisk-Standort_A
{noformat}
[Oct  1 14:51:09] VERBOSE[3029][C-00000014] chan_sip.c: Found peer 'asterisk-Standort_A' for 'asterisk-Standort_A' from 10.1.0.16:5060
{noformat}
and then we see the No compatible codecs..
{noformat}
[Oct  1 14:51:09] NOTICE[3029][C-00000014] chan_sip.c: No compatible codecs, not accepting this offer!
{noformat}
Out goes a 488 not acceptable here.

Looking at the offer and looking at the configuration for asterisk-Standort_A .. I'm not sure why Asterisk is responding the way it is. Looks unexpected certainly.