[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 15:22:10 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-26593?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233702#comment-233702 ]
Joshua Colp commented on ASTERISK-26593:
----------------------------------------
Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:
1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.
To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
As well please confirm the version of Asterisk you are using.
> 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
> 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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