[asterisk-bugs] [JIRA] (ASTERISK-26593) Switching to native rtp disables RTCP reporting
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Nov 14 11:27:10 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233693#comment-233693 ]
Asterisk Team commented on ASTERISK-26593:
------------------------------------------
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].
> 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
> 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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