[asterisk-bugs] [JIRA] (ASTERISK-20758) No audio when P2P bridge occurs
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Fri Nov 30 12:35:45 CST 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-20758?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=200304#comment-200304 ]
Joshua Colp edited comment on ASTERISK-20758 at 11/30/12 12:33 PM:
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When media is not going directly Asterisk always deconstructs the RTP packets, because the other side may not be an RTP channel. In the case of a P2P bridge since both sides are RTP the packet is minimally modified to make sure that payload numbers and such are correct. It is incredibly strange that with a jitterbuffer forced the users can talk, since implementing the P2P bridge over 5 years ago this is the first time I've ever heard of that. This may be something to do with the specific phones on each side.
was (Author: jcolp):
When media is not going directly Asterisk always deconstructs the RTP packets, because the other side may not be an RTP channel. In the case of a P2P bridge since both sides are RTP the packet is minimally modified to make sure that payload numbers and such are correct. It is incredibly strange that with a jitterbuffer forced the users can talk, since implementing the P2P bridge over 5 years ago this is the first time I've ever heard of that.
> No audio when P2P bridge occurs
> -------------------------------
>
> Key: ASTERISK-20758
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20758
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Core/RTP
> Affects Versions: 1.8.18.0, 11.0.1
> Reporter: TSAREGORODTSEV Yury
> Severity: Critical
>
> sip.conf
> [general]
> directmedia=no
> directrtpsetup=no
> [peer1]
> directmedia=no
> directrtpsetup=no
> canreinvite=no
> [peer2]
> directmedia=no
> directrtpsetup=no
> canreinvite=no
> But anyway * Send P2P packets.
> Only in case of force jitter buffer * bridging channels locally.
> But to use jitterbuffer not a right solution in this case.
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