[asterisk-bugs] [JIRA] (ASTERISK-22911) Asterisk fails to resume WebRTC call from hold

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22911?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212258#comment-212258 ] 

Matt Jordan commented on ASTERISK-22911:
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We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Please also load the res_pjsip_logger module and enable SIP trace debugging (pjsip set logger on). Please also attach your pjsip.conf. Thanks!
                
> Asterisk fails to resume WebRTC call from hold
> ----------------------------------------------
>
>                 Key: ASTERISK-22911
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22911
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_nat, Resources/res_rtp_asterisk
>    Affects Versions: SVN
>         Environment: Server:
> asterisk:svn r403157  --with-srtp --with-pjproject
> pjproject:git asterisk/pjproject HEAD --with-external-srtp --enable-shared CFLAGS="-DNDEBUG"
> Ubuntu Precise 64, 3.2.0-23-generic.
> Client:
> Chrome 33.0.1720.0 canary
> http://sipml5.org/call.htm?svn=203
>            Reporter: Vytis Valentinavičius
>
> When in call between soft-phone and WebRTC resuming from holden call does not resume the sound.
> Notices:
> 1. Hold and resume must be made by WebRTC client. Tested with sipml5.org demo.
> 2. Wireshark dump showed that after call is resumed all UDP packets do not reach WebRTC client due to wrong destination port.
> 3. Chrome stops active channel when issued hold command and creates new channel on resume. Channel is bound to new port each time.
> 4. Asterisk spits out such verbose errors:
> Before connection:
> [Nov 26 13:27:01] ERROR[2088]: pjsip:0 <?>: 	icess0x7fbe000 ..Error sending STUN request: Invalid argument
> Later in call (not related to Hold/Resume sequence):
> [Nov 26 13:28:06] WARNING[2177][C-00000000]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

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