[asterisk-bugs] [JIRA] (ASTERISK-23196) loss of voice during a call

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23196?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214678#comment-214678 ] 

Matt Jordan commented on ASTERISK-23196:
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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. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. 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
3. the behavior you actually encountered (in as much detail as possible).

This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks!




                
> loss of voice during a call
> ---------------------------
>
>                 Key: ASTERISK-23196
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23196
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 1.8.15.0
>         Environment: 2 CPU xeon E5649, 36GB RAM, CentOS release 6.2 (Final), kernel 2.6.32-220.13.1.el6.x86_64, 
> asterisk version 1.8.15-cert2
>            Reporter: Nikolay
>
> We have found out the problem with abnormal asterisk behavior during a call.
> Symptoms: One user rings to the other. The first  user is registered on one PBX, running asterisk, the second is registered on the second PBX running same version of asterisk. 
> Suddenly the the second user stops receiving voice traffic from first one - means he hears nothing, first user still hears the second one. The silence for the second user lasts for 2 second, no recovering of hearing 
> takes place. The second user then has to hang up. Very important thing here is that it happens only with second users because the use softphones. It does not happen with every call. No regularity is noticed here.
> But when we capture RTP traffic on the computers of second users we see the picture, expressed in enclosed .pcap file. Further description and consideration will concern information from this file.
> On the mark 71 (time 3,621337s ) we can see that RTP traffic is not sent from PBX to second user. However the traffic from the second user is send to PBX (that's why first user hears the second one). 
> But on the mark 182 (5,829413s), that is  2,2 seconds later the traffic suddenly appears from second  PBX to second user. Notice that we see large amount of 
> RTP packets sent from PBX to user in very short time: 0,0087 seconds - as if PBX captured those packets, that it could not send between mark 71 (time 3,621337s) and mark 182 (time 5,829413s) in some reasons, and then poured them out to user immidiately.
> Before mark 71 (time 3,621337s) and after mark 292 (time 5.838142) we can see normal mutual RTP exchange between user and PBX: three packets from user to PBX - three packets from PBX to user cyclically repeated.

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