hi,
We are currently seeing an issue where our CUBE dial-peers is configured to use TCP as the transport between a SIP call tester and the CUBE, but seems to switch to UDP when sending a BYE to end a call.
All other messaging between CUBE and call tester is done using TCP, it is only after a BYE is received from the Service Provider that a UDP BYE instead of TCP is send back to the call tester.
The endpoint (SIP call tester) then responds with a TCP BYE back after the timer expires, but as the CUBE has already cleared the call it does not recognize it and send SIP 481 back.
Has anyone come across this?
UDP BYE received from Service Provider (which is correct):
021698: Jan 18 16:27:50.219: //585583/569CD5D7B973/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxxxxxxxxx:5060;branch=z9hG4bK3612ff22c2b23e424c267570b449c179
UDP BYE sent to Call Tester (Should be TCP):
021699: Jan 18 16:27:50.220: //585582/569CD5D7B973/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:xxxxxxxx@xxxxxxxxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxxx:5060;branch=z9hG4bK84C8AFE
Call tester does not recognise the UDP BYE and sends a TCP BYE after timer ran out:
Received:
BYE sip:xxxxxxxxxx@xxxxxxxxxxxxxx:5060 SIP/2.0
Via: SIP/2.0/TCP xxxxxxxxxxxx:5060;branch=z9hG4bK-4073-1-9;rport
CUBE responds with SIP 481 using TCP again:
Sent:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/TCP xxxxxxxxxxxxxxx:5060;branch=z9hG4bK-4073-1-9;rport
Test send another TCP BYE:
Received:
BYE sip:xxxxxxxxxxx@xxxxxxxxxxxx SIP/2.0
Via: SIP/2.0/TCP xxxxxxxxxxxx:5060;branch=z9hG4bK-4073-1--1
CUBE responds again with SIP 481 using TCP:
Sent:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/TCP xxxxxxxxxxx:5060;branch=z9hG4bK-4073-1--1
Dial-Peer is confiured to use TCP:
dial-peer voice 6001 voip
description OUTBOUND to Customer PBX - Dest 2
preference 2
destination-pattern 0[1-3].........
session protocol sipv2
session target ipv4:xxxxxxxxxxxxxxxx
session transport tcp
voice-class codec 1
voice-class sip options-keepalive up-interval 10 down-interval 60 retry 3
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
no vad
!
SIP call flow is attached
