[asterisk-bugs] [JIRA] (ASTERISK-30189) chan_pjsip: rtptimeout doesn't work at all when using Stasis Application

[Sean Bright (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30189?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259970#comment-259970 ] 

Sean Bright edited comment on ASTERISK-30189 at 8/23/22 9:21 AM:
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{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}


was (Author: mikhail):
pjsip.conf

[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

> chan_pjsip: rtptimeout doesn't work at all when using Stasis Application
> ------------------------------------------------------------------------
>
>                 Key: ASTERISK-30189
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30189
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/RTP, Core/Stasis
>    Affects Versions: GIT
>            Reporter: Mikhail
>            Assignee: Unassigned
>            Severity: Major
>              Labels: webrtc
>         Attachments: debug
>
>
> 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



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