Is it possible to use session-timer (UPDATE or re-INVITE) by pjsip?

Asterisk Asterisk SIP
1

Now we are using asterisk13.27.0 and ./configure --with-pjproject-bundled for building the pjsip

When I do the incoming and outgoing call for a long time, there is no UPDATE or re-INVITE happening from our asterisk server to sip trunk. Japanese sip trunk we uses requires us to send UPDATE or re-INVITE session-timers in specific intervals. So, we need this.
Is there any bug or issues with pjsip session-timer itself?

I saw the jira issue tracking
https://issues.asterisk.org/jira/browse/ASTERISK-28003?jql=text%20~%20"pjsip%20session%20timer"
But, I can’t find the related issues.

I used this configuration in pjsip-endpoints

type = endpoint
transport=transport-udp
context=from-trunk
;allow=!all,g722,ulaw
allow=!all,ulaw		
from_domain=xxx
;force_rport=yes
;rewrite_contact=yes
;rtp_symmetric=yes
timers=required
timers_min_se=1800   
timers_sess_expires=1800 
accept_multiple_sdp_answers=yes 
direct_media=yes
connected_line_method=update 
direct_media_method=update 
direct_media_glare_mitigation=update 
disable_direct_media_on_nat=no 
force_rport=yes 
rewrite_contact=yes 
rtp_symmetric=yes
2

When I try the chan_sip, it works. So, I suspect that there is some bug or issues with pjsip.
Does anyone know any information for that?

3

I think that this is the same issue as

Using the latest asterisk didn’t solve the problem

4

I’d suggest providing an actual SIP trace of a call so it can be seen what is negotiated.

5

That is the success pattern for it. But, now, we are not sending the update packet in some reason. I don’t know why.

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6

You need to provide an actual SIP trace that can be seen (pjsip set logger on) with the contents of each packet.