[asterisk-bugs] [JIRA] (ASTERISK-26593) bridge_native_rtp: Switching to native rtp disables RTCP reporting
Joshua Colp (JIRA)
noreply at issues.asterisk.org
Mon Nov 14 15:35:10 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233704#comment-233704 ]
Joshua Colp commented on ASTERISK-26593:
----------------------------------------
The logs we need include SIP and debug level. Native RTP bridge can actually mean 2 different things. Either a local optimized one, or a remote bridge in which case media and RTCP would not be going through Asterisk. It's unclear which exactly is being done. As well the configuration would allow us to reproduce it, and different configuration influences the results of the bridge and the choice. Finally we reverted a change in Asterisk 13.2.2 which caused rtptimeout to occur when it shouldn't. Thus my question about confirmation of version.
> bridge_native_rtp: Switching to native rtp disables RTCP reporting
> ------------------------------------------------------------------
>
> Key: ASTERISK-26593
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-26593
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Bridges/bridge_native_rtp
> Affects Versions: 13.12.2
> Environment: CentOS x64
> Reporter: Luke Escude
> Assignee: Unassigned
> Severity: Minor
>
> As soon as the call switches from simple_bridge to native_rtp, the server stops receiving RTCP information, and the call will drop because it thinks there's inactivity. We've had to set the rtp timeout to 24 hours to compensate for this.
> Log readout during a phone call:
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:24 0000001231 0000000000 ( 0.00%) 0.0000 0000000779 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:24 0000000779 0000000000 ( 0.00%) 0.0000 0000000785 0000000000 ( 0.00%) 0.0023
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:25 0000001254 0000000000 ( 0.00%) 0.0000 0000000794 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:25 0000000794 0000000000 ( 0.00%) 0.0000 0000000800 0000000000 ( 0.00%) 0.0020
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:25 0000001281 0000000000 ( 0.00%) 0.0000 0000000812 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:25 0000000812 0000000000 ( 0.00%) 0.0000 0000000818 0000000000 ( 0.00%) 0.0025
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:26 0000001305 0000000000 ( 0.00%) 0.0000 0000000828 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:26 0000000828 0000000000 ( 0.00%) 0.0000 0000000834 0000000000 ( 0.00%) 0.0023
> 2 active SIP channels
> -- SIP/flowroute-000000c3 answered SIP/236-000000c2
> -- Channel SIP/flowroute-000000c3 joined 'simple_bridge' basic-bridge <f471fb8f-d8ae-4aef-b164-c35f20846fb3>
> -- Channel SIP/236-000000c2 joined 'simple_bridge' basic-bridge <f471fb8f-d8ae-4aef-b164-c35f20846fb3>
> > Bridge f471fb8f-d8ae-4aef-b164-c35f20846fb3: switching from simple_bridge technology to native_rtp
> > Locally RTP bridged 'SIP/236-000000c2' and 'SIP/flowroute-000000c3' in stack
> > Locally RTP bridged 'SIP/236-000000c2' and 'SIP/flowroute-000000c3' in stack
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:26 0000001330 0000000000 ( 0.00%) 0.0000 0000000842 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:26 0000000842 0000000000 ( 0.00%) 0.0000 0000000854 0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:28 0000001408 0000000000 ( 0.00%) 0.0000 0000000888 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:28 0000000888 0000000000 ( 0.00%) 0.0000 0000000932 0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:28 0000001434 0000000000 ( 0.00%) 0.0000 0000000906 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:28 0000000906 0000000000 ( 0.00%) 0.0000 0000000958 0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:29 0000001457 0000000000 ( 0.00%) 0.0000 0000000921 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:29 0000000921 0000000000 ( 0.00%) 0.0000 0000000981 0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:36 0000001808 0000000000 ( 0.00%) 0.0000 0000001155 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:36 0000001155 0000000000 ( 0.00%) 0.0000 0000001332 0000000000 ( 0.00%) 0.0026
> 2 active SIP channels
> vpbx36*CLI> sip show channelstats
> Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
> 216.115.69.144 4300d26f61c 00:00:37 0000001845 0000000000 ( 0.00%) 0.0000 0000001180 0000000000 ( 0.00%) 0.0001
> 98.6.78.250 1180865772- 00:00:37 0000001180 0000000000 ( 0.00%) 0.0000 0000001369 0000000000 ( 0.00%) 0.0026
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