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

Tue Feb 4 03:29:03 CST 2014

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22911?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214797#comment-214797 ] 

Igor Skomorokh edited comment on ASTERISK-22911 at 2/4/14 3:28 AM:
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Hi! I've tried 11.8.0-rc1 and can confirm, that now hold/unhold works properly but not in all cases. For example, if I initiate a call and I initiate hold and then unhold - everything works fine, but when I initiate a call and the other side accepts it an initiate hold and unhold then nothing works and Asterisk logs:

DEBUG[24993][C-00000006]: res_rtp_asterisk.c:2682 ast_rtp_write: No remote address on RTP instance '0x7f1420018308' so dropping frame
....
[Feb  4 10:04:58] DEBUG[24993][C-00000006]: channel.c:8070 ast_channel_bridge: Bridge stops bridging channels SIP/1-0000000c and SIP/2-0000000d
[Feb  4 10:04:58] DEBUG[24993][C-00000006]: res_rtp_asterisk.c:2607 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x7f1420018308'
[Feb  4 10:04:58] DEBUG[24992][C-00000006]: chan_sip.c:4568 __sip_ack: Stopping retransmission on '4ba1fb9b3575d1247dc95c6d29a375cc at 172.19.2.10:5060' of Response 64144: Match Not Found
[Feb  4 10:04:58] WARNING[24993][C-00000006]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Feb  4 10:05:00] WARNING[24993][C-00000006]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110

This might help: https://code.google.com/p/sipml5/issues/detail?id=44#c23

      was (Author: ISkomorokh):
    Hi! I've tried 11.8.0-rc1 and can confirm, that now hold/unhold works properly but not in all cases. For example, if I initiate a call and I initiate hold and then unhold - everything works fine, but when I initiate a call and the other side accepts it an initiate hold and unhold then nothing works and Asterisk logs:

DEBUG[24993][C-00000006]: res_rtp_asterisk.c:2682 ast_rtp_write: No remote address on RTP instance '0x7f1420018308' so dropping frame
....
[Feb  4 10:04:58] DEBUG[24993][C-00000006]: channel.c:8070 ast_channel_bridge: Bridge stops bridging channels SIP/1-0000000c and SIP/2-0000000d
[Feb  4 10:04:58] DEBUG[24993][C-00000006]: res_rtp_asterisk.c:2607 ast_rtp_raw_write: Starting RTCP transmission on RTP instance '0x7f1420018308'
[Feb  4 10:04:58] DEBUG[24992][C-00000006]: chan_sip.c:4568 __sip_ack: Stopping retransmission on '4ba1fb9b3575d1247dc95c6d29a375cc at 172.19.2.10:5060' of Response 64144: Match Not Found
[Feb  4 10:04:58] WARNING[24993][C-00000006]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Feb  4 10:05:00] WARNING[24993][C-00000006]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 110

> [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: Kevin Harwell
>         Attachments: capture_asterisk_211_client_15.pcap.gz, issue_22911.full.log, issue_22911.full.log, issue_22911.full.pjsip.log, works_on_my_machine.patch
>
>
> 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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