[Home]

Summary:ASTERISK-30376: Disconnecting WebRTC SIP client disconnects all other WebRTC SIP clients
Reporter:Sébastien Duthil (sduthil)Labels:webrtc
Date Opened:2022-12-23 16:40:29.000-0600Date Closed:2022-12-24 06:16:56.000-0600
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Resources/res_pjsip_pubsub
Versions:19.7.1 Frequency of
Occurrence		Constant
Related
Issues:
duplicatesASTERISK-30369 res_pjsip: Websockets from same IP shut down when they shouldn't be
Environment:Attachments:
Description:Given I have two PJSIP WebRTC endpoints A and B registered
When A disconnects
Then B is unregistered on Asterisk

I can reproduce the scenario with Asterisk 19.7.1 and the configuration given in the [WebRTC tutorial|https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5] (using two endpoints {{webrtc_client}} and {{webrtc_client2}}) and two instances of the [SIPML5 WebRTC client|https://www.doubango.org/sipml5].

The disconnection by pressing the "logout" button on SIPML5 does:
* send a SIP REGISTER with {{Contact: expires=0}}
* receives the SIP ACK
* close the SIP Websocket

Here are the logs of the WebRTC connection for both accounts:

{noformat}
 == WebSocket connection from '192.168.121.1:53748' for protocol 'sip' accepted using version '13'
   -- Added contact 'sips:webrtc_client2@192.168.121.1:53748;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0' to AOR 'webrtc_client2' with expiration of 200 seconds
 == Endpoint webrtc_client2 is now Reachable
 == WebSocket connection from '192.168.121.1:53750' for protocol 'sip' accepted using version '13'
   -- Added contact 'sips:webrtc_client@192.168.121.1:53750;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0' to AOR 'webrtc_client' with expiration of 200 seconds
 == Endpoint webrtc_client is now Reachable
{noformat}
Here is the Asterisk console when disconnecting {{webrtc_client}}:
{noformat}
   -- Removed contact 'sips:webrtc_client@192.168.121.1:53750;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0' from AOR 'webrtc_client' due to request
 == Contact webrtc_client/sips:webrtc_client@192.168.121.1:53750;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0 has been deleted
 == Endpoint webrtc_client is now Unreachable
   -- Removed contact 'sips:webrtc_client2@192.168.121.1:53748;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0' from AOR 'webrtc_client2' due to shutdown
 == Contact webrtc_client2/sips:webrtc_client2@192.168.121.1:53748;transport=ws;rtcweb-breaker=no;x-ast-orig-host=df7jal23ls0d.invalid:0 has been deleted
 == Endpoint webrtc_client2 is now Unreachable
{noformat}

We can see that {{webrtc_client2}} is disconnected as well.

We have successfully worked around this behavior in Asterisk 19.7.1 by reverting the patch for ASTERISK-30244.
Comments:By: Asterisk Team (asteriskteam) 2022-12-23 16:40:33.324-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].