[asterisk-bugs] [JIRA] (ASTERISK-26654) chan_sip: ILBC or Opus codec offer correlates with one-way audio

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26654?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234215#comment-234215 ] 

Rusty Newton commented on ASTERISK-26654:
-----------------------------------------

Looks goofy. Opening this up.  

Also, a reminder that chan_sip is under extended support since Asterisk 12, so the best way to get this issue resolved quickly is to submit a patch yourself. Otherwise, extended support modules rely on the support of the broader community as they have no guarantee of support from the core team.

If you can reproduce these issues with res_pjsip/chan_pjsip (core support) then open a new issue for that. Thanks!




> chan_sip: ILBC or Opus codec offer correlates with one-way audio
> ----------------------------------------------------------------
>
>                 Key: ASTERISK-26654
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26654
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 13.13.1
>         Environment: CentOS x64 - No NAT, public IP
>            Reporter: Luke Escude
>            Severity: Critical
>         Attachments: 2calls.pcap, asterisk_debug, extensions.conf, GOOD_AstDebug, GOODCALL.pcap, sip.conf
>
>
> We have been struggling with random (very inconsistent) one-way audio phone calls for a couple of months now. Out of the 10,000 phone calls we process per day, this only happens to less than 20 of them.
> This issue ONLY affects a few of my customers on a raw Asterisk installation, none of my FreePBX customers are having any issues.
> The server is hosted in our datacenter with a public IP - no firewall, no NAT, and it trunks to Flowroute. The customer is behind NAT, however the system works just fine like this - we always have 2-way audio between customer and our server.
> Attached is a PCAP of 2 phone calls that experienced one-way audio, and the console debug messages from asterisk.
> Flowroute is pulling a packet capture of their side for me, as we speak.
> New info: Calling her works flawlessly on FreePBX.



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