[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
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Sat May 13 00:19:58 CDT 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-26996?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236985#comment-236985 ]
Asterisk Team commented on ASTERISK-26996:
------------------------------------------
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].
> 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
>
> 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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