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

Tue Feb 18 11:28:03 CST 2014

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23213?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jonathan Rose updated ASTERISK-23213:
-------------------------------------

    Attachment: ice_candidates_excessive_clearing.patch

Hey everyone, I've created a patch (ice_candidates_excessive_clearing.patch) which I think undoes the more aggressive ICE candidate clearing from the ASTERISK-22911 patch while still addressing the holding problem that the patch was trying to fix. To put it bluntly though, I'm not especially experienced at dealing with NATS and this was really my first exposure to programming anything involved with PJNATH.

Please try this patch out and tell me if it fixes the regression. Also if the patch causes any problems, let me know if you would.

> 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: Jonathan Rose
>            Severity: Critical
>         Attachments: asterisk11-118_broke_ice.patch, extensions.conf, full_log_not_works.txt, full_log_works.txt, http.conf, ice_candidates_excessive_clearing.patch, issue_23213_BAD_asterisk_11_8_with_rtp_debug.txt, issue_23213_GOOD_astersisk_11_7_with_rtp_debug.txt, messages_log_not_works.txt, messages_log_works.txt, pjsip.conf, rtp.conf, rtp.conf, sip.conf, sip.conf, sip_peer.txt, users.conf
>
>
> 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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