[asterisk-bugs] [JIRA] (ASTERISK-26996) choppy sound: asterisk can't decide which codec to use. Request an option to force asterisk always to use one codec in ok SDP and not a list

Joshua Elson (JIRA) noreply at issues.asterisk.org
Wed May 17 16:54:57 CDT 2017


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26996?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237028#comment-237028 ] 

Joshua Elson commented on ASTERISK-26996:
-----------------------------------------

We've certainly seen issues on this as well. While the root cause here is usually a poorly implemented peer, issues like this happen across many different endpoints, it would probably make sense to have a chan_sip equivalent option like preferred_codec_only that would just respond with only the most preferred codec in the OK.

Just a thought...

> choppy sound: asterisk can't decide which codec to use. Request an option to force asterisk always to use one codec in ok SDP and not a list
> --------------------------------------------------------------------------------------------------------------------------------------------
>
>                 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
>         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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