[asterisk-bugs] [JIRA] (ASTERISK-23213) SIP over WS: Audio problems when upgrading to 11.8 from 11.7 with endpoints behind NAT

Fri Feb 7 08:45:03 CST 2014

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23213?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214923#comment-214923 ] 

David Brillert commented on ASTERISK-23213:
-------------------------------------------

So your patch reverts this code
http://svnview.digium.com/svn/asterisk?view=revision&revision=405234

Revision 405234 - Directory Listing
Modified Thu Jan 9 16:49:55 2014 UTC (4 weeks ago) by kharwell

res_rtp_asterisk: Fails to resume WebRTC call from hold

In ast_rtp_ice_start if the ice session create check list failed, start check
was never initiated and ice_started was never set to true.  Upon re-entering
the function (for instance, [un]hold) it would try to create the check list
again with duplicate remote candidates.

Fixed so that if the create check list fails the necessary data structures
are properly re-initialized for any subsequent retries.

Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a
check list failure because it possible things might still work.  However, a
debug message was added to help with any future troubleshooting.

(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Patches:
     works_on_my_machine.patch uploaded by xytis (license 6558)

> SIP over WS: Audio problems when upgrading to 11.8 from 11.7 with endpoints behind NAT
> --------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23213
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23213
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.8.0
>         Environment: Ubuntu 12.04 LTS 64 bit. 
> JsSIP
> WebRTC
>            Reporter: Andrea Suisani
>            Assignee: Matt Jordan
>            Severity: Critical
>         Attachments: asterisk11-118_broke_ice.patch, issue_23213_BAD_asterisk_11_8_with_rtp_debug.txt, issue_23213_GOOD_astersisk_11_7_with_rtp_debug.txt, rtp.conf, sip.conf, sip_peer.txt
>
>
> Hi all, 
> We're testing the 11.8.0-rc1 with the aim to deploy it on our production servers. During this test we've found out that for some strange reason the audio of our phone calls stop working. Neither the callee nor the caller here anything. 
> I want briefly describe our setup. We're using a web soft-phone developed using JsSIP and WebRTC. Such application is connected to asterisk using websocket, than asterisk routes the call to the endpoint using a VoIP SIP provider.
> Both the astersik server and the webapp are on the same LAN, and both are connected to the internet through NAT. 
> Everything works as expected as long as we use Asterisk <= 11.7.0, as soon as we move to 11.8.0-rc1 the audio simply goes away. 
> I can reproduce the problem at will. 
> I've looked at the rtp debug output both for a working session (11.7)  and a not working session (11.8). In the former case we have something like: 
> {noformat}
> Got  RTP packet from    80.xxx.xxx.xxx:10244 (type 18, seq 042964, ts 084000, len 000020)
> Sent RTP packet to      192.168.1.50:44635 (via ICE) (type 00, seq 049633, ts 094560, len 4294967284)
> Got  RTP packet from    80.xxx.xxx.xxx:10244 (type 18, seq 042965, ts 084160, len 000020)
> Sent RTP packet to      192.168.1.50:44635 (via ICE) (type 00, seq 049634, ts 094720, len 4294967284)
> Got  RTP packet from    80.xxx.xxx.xxx:10244 (type 18, seq 042966, ts 084320, len 000020)
> Sent RTP packet to      192.168.1.50:44635 (via ICE) (type 00, seq 049635, ts 094880, len 4294967284)
> {noformat}
> whereas in the latter we got: 
> {noformat}
> Got  RTP packet from    80.xxx.xxx.xxx:15456 (type 18, seq 062132, ts 065600, len 000020)
> Sent RTP packet to      78.zzz.zzz.zzz:43994 (type 00, seq 024953, ts 073760, len 000170)
> Got  RTP packet from    80.xxx.xxx.xxx:15456 (type 18, seq 062133, ts 065760, len 000020)
> Sent RTP packet to      78.zzz.zzz.zzz:43994 (type 00, seq 024954, ts 073920, len 000170)
> Got  RTP packet from    80.xxx.xxx.xxx:15456 (type 18, seq 062134, ts 065920, len 000020)
> {noformat}
> as you might see when the audio works properly, the rtp connection is established between the IP address of our SIP VoIP provider (80.xxx.xxx.xxx) and IP address of the dev box 192.168.1.50 where the webapp is running (via ICE). 
> On the other case using asterisk 11.8  the audio does not flow properly because one of the end-point is the box associated with public IP address configured  on the router that is NATting all the outgoing traffic of the development LAN (78.zzz.zzz.zzz), instead of being the IP of the dev box from which we issued the call first place.
> edit 1: I forgot to mention that I've used the same exact configuration both for asterisk 11.7 and 11.8-RC1

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