PJSIP call finish event on Webrtc only after 30 seconds

Hi guys!
Asterisk version 17.5.1
I’m migrating to pjsip from chan_sip, and now i face a problem then client hangs up a line, webrtc client get it only after 30-45 seconds from event
If i’m finishing a call - everything is ok - i get end event immidiatly, strange thing - that it should be?
Thanks in advance

Please provide the protocol debugging logs from chan_pjsip.

00002 1593263359 * <== AB.AB.AB.ABC:65358 INVITE sip:XXXXXXXXXX@ip.mysystem.com SIP/2.0
00003 1593263359 * ==> AB.AB.AB.ABC:65358 SIP/2.0 401 Unauthorized
00004 1593263359 * <== AB.AB.AB.ABC:65358 ACK sip:XXXXXXXXXX@ip.mysystem.com SIP/2.0
00005 1593263359 * <== AB.AB.AB.ABC:65358 INVITE sip:XXXXXXXXXX@ip.mysystem.com SIP/2.0
00006 1593263359 * ==> AB.AB.AB.ABC:65358 SIP/2.0 100 Trying
00007 1593263359 * ==> BC.BC.BC.BCD:5060 INVITE sip:XXXXXXXXXX@ip.provider.com SIP/2.0
00008 1593263359 * <== BC.BC.BC.BCD:5060 SIP/2.0 100 Trying
00009 1593263359 * <== BC.BC.BC.BCD:5060 SIP/2.0 183 Session Progress
00010 1593263359 * ==> BC.BC.BC.BCD:5060 PRACK sip:xxx.xxx.xxx.ip.provider.com:5060 SIP/2.0
00011 1593263359 * ==> AB.AB.AB.ABC:65358 SIP/2.0 183 Session Progress
00012 1593263359 * ==> AB.AB.AB.ABC:65358 SIP/2.0 183 Session Progress
00013 1593263359 * <== BC.BC.BC.BCD:5060 SIP/2.0 200 OK
00014 1593263360 * <== BC.BC.BC.BCD:5060 SIP/2.0 200 OK
00015 1593263360 * ==> BC.BC.BC.BCD:5060 ACK sip:xxx.xxx.xxx.ip.provider.com:5060 SIP/2.0
00016 1593263360 * ==> AB.AB.AB.ABC:65358 SIP/2.0 200 OK
00017 1593263360 * <== AB.AB.AB.ABC:65358 ACK sip:XX.XX.XXX.XXX:5060 SIP/2.0
00018 1593263367 * <== BC.BC.BC.BCD:5060 BYE sip:XXXXXXXX@XX.XX.XXX.XXX:5060 SIP/2.0
00019 1593263367 * ==> BC.BC.BC.BCD:5060 SIP/2.0 200 OK
00020 1593263367 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00021 1593263368 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00022 1593263369 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00023 1593263371 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00026 1593263375 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00027 1593263379 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00032 1593263383 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00037 1593263387 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00038 1593263391 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00039 1593263395 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00040 1593263399 * ==> AB.AB.AB.ABC:65358 BYE sip:g9249403@AB.AB.AB.ABC:65358;transport=ws;ob SIP/2.0
00043 1593263405 * <== AB.AB.AB.ABC:65358 UPDATE sip:XX.XX.XXX.XXX:5060 SIP/2.0
00044 1593263405 * ==> AB.AB.AB.ABC:65358 SIP/2.0 481 Call/Transaction Does Not Exist
00045 1593263405 * <== AB.AB.AB.ABC:65358 BYE sip:XX.XX.XXX.XXX:5060 SIP/2.0
00046 1593263405 * ==> AB.AB.AB.ABC:65358 SIP/2.0 481 Call/Transaction Does Not Exist

I get call end only after 50-60 seconds on webrtc client -
In asterisk cli i see hangup - and no messages more

This is only a summary, not the full log.

The call was cleared by BCD after 7 seconds.

ABC ignored Asterisk’s attempts to clear the ABC side of the call

Asterisk eventually gave up.

ABC did an UPDATE for reasons that will only be clear from the full log, but it was pointless because Asterisk has already given up even trying to close the call normal. It looks like ABC has stopped seeing anything sent by Asterisk.

How i can show full log?
If i use chan_sip - everything ok - if i finish call from webrtc everything ok - it seems something with PJSIP?

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information#CollectingDebugInformation-Enablechanneltechorfeaturespecificdebug

i get 300kb log - may be i can specify only one call log?

Thanks david551, i found that i use not same transport as on chan_sip. I set it to udp, but in real i’m in wss. If i change it - everything good now! Thank you!

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