[asterisk-bugs] [JIRA] (ASTERISK-24002) No audio after WebRTC callee resumes call from hold

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24002?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=220419#comment-220419 ] 

Joshua Colp commented on ASTERISK-24002:
----------------------------------------

Ah, it's a SipML5 patch to force SDES. Even then... going forward DTLS-SRTP is the only way to go. Any time spent on SDES in WebRTC is time wasted as SDES will soon not exist for it.

> No audio after WebRTC callee resumes call from hold
> ---------------------------------------------------
>
>                 Key: ASTERISK-24002
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24002
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Channels/chan_sip/SRTP, Core/RTP
>    Affects Versions: 12.2.0, 12.3.0, 12.4.0
>         Environment: Ubuntu 14.04
> PJProject(https://github.com/asterisk/pjproject 06/JUN/14)
> --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp CFLAGS="-g -DNDEBUG"
> chromium/chrome M35 (debug launch options: --disable-user-media-security --enable-logging --v=4 --vmodule=*libjingle/source/talk/*=4 --vmodule=*media/audio/*=4)
> SIPml-api.js?svn=224 with SDES patch from https://code.google.com/p/sipml5/issues/detail?id=183
>            Reporter: Aleksei Kulakov
>            Assignee: Aleksei Kulakov
>         Attachments: calleeChromeConsole.txt, calleeChromeWebrtcDump.txt, extensions.conf, http.conf, noAudioAfterWebRtcCalleeUnhold.log, noAudioAfterWebRtcCalleeUnhold.pcapng, pjsip.conf
>
>
> # Caller 6001(SIP softphone, but results for WebRTC or 'originate' caller is the same) calls WebRTC endpoint 354(SIPml)
> # Callee places call on hold
> # Callee resumes call from hold
> After resuming there is no audio on both ends and many of 'SRTP unprotect failed with: authentication failure 110' messages in asterisk log.
> _NB: When endpoints in pjsip.conf configured to use DTLS, resuming call from hold fails with '488 Not acceptable here'_
> Issue reproducible with chan_pjsip(logs for that case) and chan_sip.
> Can be reproduced even when call is originated by asterisk('originate PJSIP/354 application musiconhold')
> It is related to [ASTERISK-19609]:
> When resuming from hold Chrome sends packet  with 2 crypto lines:
> {quote}
> a=group:BUNDLE audio
> a=crypto:0 AES_CM_128_HMAC_SHA1_32
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> {quote}
> And asterisk replies with 
> bq. a=crypto:1 AES_CM_128_HMAC_SHA1_80
> Then Chrome starts spamming log with mesages like
> {quote}
> [24:87:0709/161528:VERBOSE2:srtpfilter.cc(535)] Failed to unprotect SRTP packet, err=7
> [24:87:0709/161528:VERBOSE1:channel.cc(575)] Failed to unprotect audio RTP packet: size=176, seqnum=38058, SSRC=699584917
> {quote}
> After adding "srtp_tag_32=yes" option to target endpoint(pjsip.conf), Asterisk wont spam "SRTP Unprotect failed" errors and Chrome logs following:
> {quote}
> [24:87:0709/164658:VERBOSE2:srtpfilter.cc(365)] Invalid parameters in SRTP answer
> ...
> [24:24:0709/164658:VERBOSE1:webrtcsession.cc(251)] Failed to set remote answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to setup SRTP filter..
> {quote}
> and start spamming log with
> {quote}
> [24:87:0709/164658:VERBOSE1:port.cc(296)] Jingle:Port[audio:1:0::Net[eth0:192.168.0.139/32]]: Received non-STUN packet from unknown address (192.168.0.139:10086)
> {quote}
> *So it looks like that something went wrong with crypto negotiation procedure.*
> *Workaround: disable media encryption altogether:*
> * Remove 'media_encryption' option from endpoint in pjsip.conf
> * Run chrome with --disable-webrtc-encryption (should work only in dev builds, but chromium M35 on xubuntu 14.04 accepts it too)



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