[asterisk-bugs] [JIRA] (ASTERISK-28342) Ast-to-Ast setup using the same rtcpinterval crashes RTCP and audio stream

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28342?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246610#comment-246610 ] 

Asterisk Team commented on ASTERISK-28342:
------------------------------------------

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].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

> Ast-to-Ast setup using the same rtcpinterval crashes RTCP and audio stream
> --------------------------------------------------------------------------
>
>                 Key: ASTERISK-28342
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28342
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 16.2.1
>         Environment: CentOS 7 (7.6-1810 x86_64)
>            Reporter: Speed Dial Dave
>            Severity: Minor
>              Labels: pjsip
>
> When stress-testing PJSIP and Asterisk 16.2.1 by doing direct calls between two identically configured Asterisk setups (over both WAN and LAN) I noticed a specific error message that kept showing up in the logs:
> res_rtp_asterisk.c: RTCP SR transmission error to 1.2.3.4:12345, rtcp halted Operation not permitted
> This started showing up when my tests reached about 200 simultaneously connected calls between the two Asterisk setups. At this level, after a total of 15k calls, the error showed up about 70-80 times on the Asterisk doing the dialing, and about 10-15 times on the one receiving the calls; at ~350 simultaneous calls the number of errors was much higher. When this error popped up the outgoing audio stream ("us") also dropped out from the affected calls, but the channels didn't crash and the calls kept going and finalized normally without any other apparent error.
> As "operation not permitted"/EPERM should be a permission error I first fiddled around with increasing various ulimits as I thought it was some resource starvation, but without luck. After experimenting with various config changes for Asterisk I at one point had different rtcpinterval (rtp.conf) values on the two machines - one was set to 3000 and the other to 6000 - and this completely cleared the problem. Running my test several times more with the two machines having an rtcpinterval difference of only 500 I couldn't get the error to show up again, even when stepping it up to 600 simultaneous calls.



--
This message was sent by Atlassian JIRA
(v6.2#6252)