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

Luke Escude (JIRA) noreply at issues.asterisk.org
Mon Dec 19 18:02:10 CST 2016


     [ https://issues.asterisk.org/jira/browse/ASTERISK-26654?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Luke Escude closed ASTERISK-26654.
----------------------------------

    Resolution: Fixed

Will start a new ticket regarding PJSIP and Opus codec offer.

> 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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