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

[Vytis Valentinavičius (JIRA)]

Vytis Valentinavičius created ASTERISK-22911:
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             Summary: 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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