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

[r mundkowsky (JIRA)]

r mundkowsky created ASTERISK-24979:
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             Summary: 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: 13.3.2, 13.2.0
         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).



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