Trying to test 302 Temporarily moved
Paul Whitfield
Hi I am trying to test the following scenario.
UA sends an INVITE
SIP Inspector 302 Moved Temporarily (with new contact details)
UA Ack
UA Sends an INVITE with new contact details
SIP Inspector 100
SIP Inspector 180
SIP Inspector 200
However when I try to set-up this scenario, I get locked in a loop where the second Invite takes me back to the top of the scenario.
How can I prevent this?
Regards
Paul
Scenario file follows:
-----
<----------
INVITE
---------->
SIP/2.0 302 Moved Temporarily
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
User-Agent: SIPInspector_v_2.40
Contact: <sip:9998@[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
<----------
ACK
<----------
INVITE
---------->
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
User-Agent: SIPInspector_v_2.00
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
---------->
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
User-Agent: SIPInspector_v_2.00
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
---------->
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=q45243sgfsagtfwqt
[last_Call-ID:]
[last_CSeq:]
User-Agent: SIPInspector_v_2.00
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=111 843670094 843670094 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
t=0 0
a=sendrecv
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
<----------
ACK
----------
cmd_PlayPcap=.\pcap\test.pcap
<----------
BYE
---------->
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
User-Agent: SIPInspector_v_2.00
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
SIP Inspector
Hi Paul!
Do both INVITEs have the same value under Call-ID header? If they don't then that would be reason for this behavior. Can you clarify please?
Zarko
Paul Whitfield
Hi Zarko,
That is the problem – the second INVITE does have a new Call-ID, I will fix that and see how I go them…
Regards
Paul Whitfield.
Zarko C
It appears you are working on a UAC and use SIP Inspector to simulate the server side. In that case UAC does not need to use the same Call-ID value in both INVITE requests.
To achieve wanted behavior of the server you can have 2 SIP Inspector instances running at the same IP address but different ports. One at IP:port1 and the other at IP:port2. The one at IP:port1 handles initial INVITE and redirects call to IP:port2 instance of SIP Inspector where the 2nd INVITE and a brand new call is handled all together. Just a suggestion how to organize your setup.
At the same time I don't why 2nd INVITE must have a different Call-ID. I believe the real life servers do not care whether Call-ID values in INVITE before and after redirect are the same or not.
Zarko