[asterisk-bugs] [JIRA] (ASTERISK-20098) P2P bridging can cause the SSRC of a RTP session to change during a call
Torrey Searle (JIRA)
noreply at issues.asterisk.org
Thu Oct 31 03:42:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-20098?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211374#comment-211374 ]
Torrey Searle commented on ASTERISK-20098:
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>From my personal viewpoint we have had customer complains of one way audio when srcc changes (I don't have the equipment references, seems locking onto an ssrc is a common practice)
I have some comments with regards to this patch:
1. this patch adds a config option to disable p2p bridge.. there are already cases already where that would happen e.g. if you need to convert between DTMF INFO and RFC2833 this patch cannot introduce a bug that doesn't already exist
2. RTP frames include timing information in them. So there should no jump in timestamps between the early bridge and the native bridge afterwards
As a more generic solution may I suggest the following:
in ast_rtp_read move the SSRC change detection logic to be just before bridge_p2p_rtp_write (instead of just after)
modify bridge_p2p_rtp_write to re-write the ssrc and seqno of the rtp struct already has a ssrc defined (timing shouldn't need to be re-written as it was never-rewritten)
What do you guys think?
> 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.
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