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

[Vytis Valentinavičius (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22911?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212495#comment-212495 ] 

Vytis Valentinavičius edited comment on ASTERISK-22911 at 12/4/13 2:22 PM:
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After a while I think I have found a solution:
During second (and consequent) SIP INVITE, which negotiates new port for audio channel ICE session remote candidates are not updated. This had to be fixed by removing flag which prevented re-pairing of remote candidates and by clearing RTP session container, which held SIP INVITE announced ICE candidates after pairing.

I can submit a patch for these changes. I specifically targeted 11.5.1, since it is my priority target.
Where should I submit this patch?
                
      was (Author: xytis):
    After a while I think I have found a solution:
During second (and consequent) SIP INVITE, which negotiates new port for audio channel ICE session remote candidates are not updated. This had to be fixed by removing flag which prevented re-pairing of remote candidates and by clearing RTP session container, which held SIP INVITE announced ICE candidates after pairing.

I can submit a patch for these changes. I specifically targeted 11.5.1, since it is my priority target.
Where should I submit this patch?
                  
> 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, Resources/res_rtp_asterisk
>    Affects Versions: 12.0.0-beta2
>         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
>            Assignee: Vytis Valentinavičius
>         Attachments: capture_asterisk_211_client_15.pcap.gz, issue_22911.full.log, issue_22911.full.log, issue_22911.full.pjsip.log
>
>
> 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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