[asterisk-bugs] [JIRA] (ASTERISK-26996) chan_pjsip: Flipping between codecs

[George Joseph (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26996?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

George Joseph updated ASTERISK-26996:
-------------------------------------

    Target Release Version/s: 14.6.0

> chan_pjsip: Flipping between codecs
> -----------------------------------
>
>                 Key: ASTERISK-26996
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26996
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.14.1, 13.15.0
>         Environment: CentOS 6, linux 64bit 
>            Reporter: Michael Maier
>            Assignee: Joshua Colp
>      Target Release: 13.17.0, 14.6.0
>
>         Attachments: conference-broken-codec choice
>
>
> Asterisk initiates a call and provides more than one codec in SDP (e.g. g722, alaw, ...). The callee accepts the list of codecs in ok SDP (g722, alaw).
> At this point, asterisk isn't always able to decide which codec to use later on. It frequently switches between g722 and alaw which leads to choppy sound.
> But that's not always happening - it's showing up here, if the callee sends the first rtp pacakge - the problem seams not to happen, if the caller sends the first rtp package (g722) - I did multiple tests.
> The attached debug output is an example of the broken situation, if asterisk can't decide what codec to use.
> Goal is, to ensure, that asterisk always uses the primary codec of ok SDP list, or, maybe the better solution, just to provide one codec in SDP ok and not a list. Maybe there are other devices out there, which do have similar problems.



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