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

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24979?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227589#comment-227589 ] 

Rusty Newton commented on ASTERISK-24979:
-----------------------------------------

That is not going to work out. Please provide packet captures and debug logs from your system (running Asterisk 13.5.0 or the latest head of the Git 13 branch). We can't reproduce the issue at the moment but we can take a look to see if there is anything obvious in the logs or pcap. Other than that with only one report of this issue and no way to reproduce it on a fresh system with the latest version we'll have to close this out until we get more information.

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
>            Assignee: Unassigned
>         Attachments: full-ips-removed.txt
>
>
> Steps that Cause error:
> 	
> webrtc client (WEBRTC_PUBLIC_IP using sipml5 or jssip via user 6002) makes audio call to extension 4000 on Asterisk (ASTRISK_PUBLIC_IP) via ws protocol directly or via webrtc2sip.  Asterisk answers, plays local audio file and dials SIP/4000 at JVXMLcloud1 peer (BROWSER_IP). SIP/4000 at JVXMLcloud1 peer then starts dialog with webrtc client. SIP/4000 at JVXMLcloud1 peer handles SIP traffic, but hands off RTP to sound server (SOUNDSRV_IP). webrtc client hangs up.
> Expected behavior:
> webrtc client has dialog with SIP/4000 at JVXMLcloud1 peer.  All audio sent from SIP/4000 at JVXMLcloud1 is heard by webrtc client.
> Actual behavior:
> webrtc client connects to Asterisk and hears Asterisk played audio file then hears some small parts of dialog that is expected from SIP/4000 at JVXMLcloud1 peer. webrtc client hangs-up before dialog is finished because no audio is heard after awhile.
> Notes:
> 1) The audio logs on Asterisk though have the complete audio to/from SIP/4000 at JVXMLcloud1, nothing is missing.  
> 2) Output audio sent to webrtc client is discontinuous (RTP send “hello”, pause (no RTP), …).  
> 3) Normal SIP/RTP clients (such as Noiper, Peers, PSTN/SIP) work correctly hearing full dialog from SIP/4000 at JVXMLcloud1 peer.  
> 4) Output to the webrtc client is fine if the webrtc client is called from a SIP/RTP client (such as Noiper, Peers) via Asterisk server.	
> sip.conf snippet:
> {noformat}
> [6002]
> transport=ws
> defaultuser=6002
> host=dynamic
> secret=AAAAAAA
> type=friend
> encryption=yes
> avpf=yes
> force_avp=yes
> icesupport=yes
> directmedia=no
> hasiax=no
> hassip=yes
> disallow=all
> allow=ulaw
> allow=g729
> allow=g723
> allow=speex
> allow=ilbc
> allow=gsm
> dtlsenable=yes
> dtlsverify=fingerprint
> dtlscertfile=/export/Apps/asterisk/lib/asterisk/keys/asterisk.pem
> dtlscafile=/export/Apps/asterisk/lib/asterisk/keys/ca.crt
> dtlssetup=actpass
> [JVXMLcloud1]
> type=peer
> nat=no
> qualify=no
> host=BROWSER_IP
> port=5090
> dtmfmode=info
> canreinvite=no
> {noformat}
> extensions.conf snippet:
> {noformat}
> [system]
> exten => 4000,1,Answer()
> exten => 4000,n,Playback(/export/Apps/asterisk13-3-2/var/lib/asterisk/sounds/en/custom/transferring_with_record_warning)
> exten => 4000,n,NoOp(Caller ID: ${CALLERID(num)})
> exten => 4000,n,NoOp(SIP Headers:: From: ${SIP_HEADER(From)}, To: ${SIP_HEADER(To)})
> exten => 4000,n,NoOp(Unique ID: ${UNIQUEID}, Channel Name: ${CHANNEL}, SIP Call ID: ${SIPCALLID})
> exten => 4000,n,NoOp(The current group count : ${GROUP_COUNT(outgoing)})
> exten => 4000,n,GotoIf($[${GROUP_COUNT(outgoing)}>1]?ring)
> exten => 4000,n,Set(GLOBAL(current_timestamp_4000)=${STRFTIME(${EPOCH},GMT+1,%s)})
> exten => 4000,n,Set(GLOBAL(current_full_format_timestamp_4000)=${STRFTIME(${EPOCH},GMT-8,%d%m%Y_%H%M%S)})
> exten => 4000,n,NoOp(The current timestamp : ${current_timestamp_4000})
> exten => 4000,n,NoOp(The last timestamp : ${last_timestamp_4000})
> exten => 4000,n,GotoIf($[(${last_timestamp_4000}+20>${current_timestamp_4000})]?ring)
> exten => 4000,n,NoOp(All cases passed)
> exten => 4000,n(ring),Ringing()
> exten => 4000,n,Wait(2)
> exten => 4000,n,Monitor(wav,SYSTEM_audio_ext_4000_${current_full_format_timestamp_4000})
> exten => 4000,n,MixMonitor(SYSTEM_MixMonitor_audio_ext_4000_${current_full_format_timestamp_4000}.wav)
> exten => 4000,n,NoOp(The full call corresponding to SIP Call ID: ${SIPCALLID} and Caller ID: ${CALLERID(num)} was recorded to the file: SYSTEM_MixMonitor_audio_ext_4000_${current_full_format_timestamp_4000}.wav)
> exten => 4000,n,Dial(SIP/4000 at JVXMLcloud1,,XgF(default^4000^14))
> exten => 4000,n,Set(GLOBAL(last_timestamp_4000)=${STRFTIME(${EPOCH},GMT+1,%s)})
> exten => 4000,n,Hangup()
> exten => 4000,n,Ringing()
> exten => 4000,n,Wait(2)
> exten => 4000,n,Playback(/export/Apps/asterisk/lib/asterisk/sounds/custom/busy_later)
> exten => 4000,n,Wait(1)
> exten => 4000,n,Hangup()
> {noformat}
> rtp.conf:
> {noformat}
> [general]
> rtpstart=10000
> rtpend=20000
>                         ;(min 500, max 60000, default 5000)
> icesupport=true
> stunaddr=stun.l.google.com:19302
> {noformat}
> http.conf:
> {noformat}
> [general]
> enabled=yes
> bindaddr=0.0.0.0
> bindport=8088
> {noformat}



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