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Summary:ASTERISK-30189: chan_pjsip: rtptimeout doesn't work at all when using Stasis Application
Reporter:Mikhail (Mikhail)Labels:webrtc
Date Opened:2022-08-23 08:07:31Date Closed:2022-09-13 12:00:33
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/RTP Core/Stasis
Versions:GIT Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debug
Description:Setting rtptimeout in PJSIP Conf is supposed to terminate the call when no RTP Information was received after a certain time

Test Procedure:
   Setup an extension throwing into a StasisApp
   Have the StasisApp make the channel join a bridge
   Abruptly cut the PJSIP/RTP client through a SIGKILL or network connectivity loss

Expected Result:
Asterisk detects the lack of RTP traffic and terminates the call after the set timeout, notifying in Console, and the ARI Application via StasisEnd/ChannelLeftBridge/ChannelDestroyed

Actual Result:
Nothing happens, call goes on despite receiving no data
Comments:By: Asterisk Team (asteriskteam) 2022-08-23 08:07:37.681-0500

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.

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By: Joshua C. Colp (jcolp) 2022-08-23 08:12:15.788-0500

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


By: Joshua C. Colp (jcolp) 2022-08-23 08:12:58.650-0500

Additionally, provide the actual configuration. And have you confirmed that it works when not in the Stasis application?
By: Mikhail (Mikhail) 2022-08-23 08:23:42.919-0500

{code:title=pjsip.conf|borderStyle=solid}
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
; All other transport parameters are ignored for wss transports.

[udp-transport]
type=transport
protocol=udp
bind=0.0.0.0

[webrtc_task]
type=aor
max_contacts=100
remove_existing=no
 
[webrtc_task]
type=auth
auth_type=userpass
username=webrtc_task
password=webrtc_task

[webrtc_task]
rtp_timeout=10
ice_support=yes
rtcp_mux=yes
type=endpoint
aors=webrtc_task
auth=webrtc_task
dtls_auto_generate_cert=yes
webrtc=yes
context=task_service
disallow=all
allow=alaw
{code}

{code:title=extensions.conf|borderStyle=solid}
[task_service] -- rtp_timeout not working
exten => _X.,1,NoOp(Call from ${CALLERID(num)} for task ${EXTEN})
same => n,Stasis(TaskServiceApp,callId:${PJSIP_HEADER(read,Call-ID)})
same => n,Hangup()

[task_service_test] -- rtp_timeout work
exten => _X.,1,dial(PJSIP/Tom,10)
{code}
By: Joshua C. Colp (jcolp) 2022-08-23 08:36:37.825-0500

Was the channel actually answered in both scenarios (thus they are actually equivalent)? In the debug provided it doesn't look like it was.
By: Mikhail (Mikhail) 2022-08-23 08:48:27.439-0500

for two scenarios: call accepted and call not accepted
By: Benjamin Keith Ford (bford) 2022-08-29 14:16:19.716-0500

Just tested this and it seems to disconnect the call after terminating network connection. A {{StasisEnd}} event is also triggered. Can you provide your Stasis application to see if there's a step that you might have that could cause the issue?
By: Asterisk Team (asteriskteam) 2022-09-13 12:00:32.008-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines