[asterisk-bugs] [JIRA] (ASTERISK-20098) P2P bridging can cause the SSRC of a RTP session to change during a call

[Andre Di Genova (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20098?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214292#comment-214292 ] 

Andre Di Genova edited comment on ASTERISK-20098 at 1/21/14 10:34 PM:
----------------------------------------------------------------------

The changing of SSRCs on bridging also causes problems with Google's implementation of WebRTC. The changing of SSRCs results in loss of audio being sent by Asterisk.

Internally I think they use SSRC to keep track of audio streams in libjingle.
                
      was (Author: AndrePew):
    The changing of SSRCs on bridging also causes problems with Google's implementation of WebRTC. The changing of SSRC's results in loss of audio being sent by Asterisk.

Internally I think they use SSRC to keep track of audio streams in libjingle.
                  
> P2P bridging can cause the SSRC of a RTP session to change during a call
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-20098
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20098
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Core/RTP
>            Reporter: Morten Tryfoss
>            Severity: Minor
>         Attachments: p2pbridge.diff
>
>
> P2P bridging can cause the SSRC id for an RTP stream to change during an active call. This can make some equiptment unhappy.
> I've added a configuration option so that this can be turned off.
> See the attached patch.

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira