[asterisk-bugs] [JIRA] (ASTERISK-24979) Webrtc client audio output is consistently skipping or missing non-continuous audio

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24979?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225942#comment-225942 ] 

Matt Jordan commented on ASTERISK-24979:
----------------------------------------

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



> Webrtc client audio output is consistently skipping or missing non-continuous audio
> -----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24979
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24979
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Resources/res_http_websocket
>    Affects Versions: 13.2.0, 13.3.2
>         Environment: Server: Ubuntu 14.04.1 LTS x86_64 
> web sockets server: Asterisk or webrtc2sip
> clients: sipml5 and jssip
>            Reporter: r mundkowsky
>
> When using webrtc client to call Asterisk directly or via webrtc2sip, audio output to the webrtc client is consistently missing audio. The audio logs on Asterisk though have the complete audio with nothing missing.  Note that Asterisk is connecting the client to 2 servers on the backend.  The first  server JXVML handles sip and hands RTP off to another server.  Also note that the output audio is discontinuous (RTP send “hello”, pause (no RTP), …).  Also note that normal SIP/RTP clients (such as Noiper, Peers) work correctly calling this same extension.  Also note that output to the webrtc client is fine if the webrtc client is called from a SIP/RTP client (such as Noiper, Peers).



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