I'm not sure I understand what you are asking for.
A "call" in a scenario actually represents any single SIP dialog, which
can currently be established using an INVITE and terminated using a BYE,
or established using a SUBSCRIBE and terminated using another SUBSCRIBE
or a NOTIFY, or established using a REFER and terminated using a NOTIFY.
The state of each dialog changes independently even if the same UA
participates in both. So making a SIPp scenario or an actual UA permit
messages belonging to two independent dialogs affect a state machine
common for both would not be a good idea as that state machine would
anyway have to be two in one.
So describe in more detail the expected behavior, there are ways to
synchronize two dialogs using additional SIPp instances, so that you
could e.g. initiate a pickup call by sending an INVITE with Replaces
when you receive a NOTIFY about an incoming call to another UA.
Pavel
Dne 29.6.2018 v 16:11 Mert Metin napsal(a):
Hi,
We try to add presence to our sipp xml’s.
We can send subscribe and receive notify .
When subscriber status has changed one notify message comes to sipp
different Call-ID with invite.
Different Call-ID’s gives error by sipp. How can we implement
different Call-ID’s in one call scenario?
We can see that these two message are two different call by sipp.
But we want to run sipp with different two Call-ID in one call scenario.
NOTIFY Call-ID is: [email protected]
<mailto:[email protected]>
INVITE Call-ID is: [email protected]
<mailto:[email protected]>
NOTIFY sip:[email protected]
<mailto:sip%[email protected]>;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.179.153:5060;branch=z9hG4bKfBzr2tL
Max-Forwards: 70
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=2a7847d46
To: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=415684934
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>
Call-ID: [email protected] <mailto:[email protected]>
CSeq: 57049 NOTIFY
User-Agent: Sip-Spc v. t111 d65535
Supported: replaces, timer
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,PRACK,UPDATE,PUBLISH,SUBSCRIBE
Event: dialog
Subscription-State:active
Content-Type: application/dialog-info+xml
Content-Length: 208
<?xml version='1.0'?>
<dialog-info xmlns='urn:ietf:params:xml:ns:dialog-info' version='0'
state='full'entity='sip:[email protected]
<mailto:sip%[email protected]>'>
<dialog id='15000'>
<state>confirmed</state>
</dialog>
</dialog-info>
INVITE sip:[email protected]
<mailto:sip%[email protected]>;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.179.153:5060;branch=z9hG4bKW37cJ1p
Max-Forwards: 70
From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=896c33429
To: <sip:[email protected] <mailto:sip%[email protected]>>
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>
Call-ID: [email protected]
<mailto:[email protected]>
CSeq: 125930 INVITE
Session-Expires: 300;refresher=uac
Min-SE: 90
User-Agent: Sip-Spc v. t47 d49
Supported: replaces, 100rel, timer
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,PRACK,UPDATE,PUBLISH,SUBSCRIBE
Alert-Info:<http://192.168.179.153:3802/internal_ring_tone.wav>;info=Internal
Content-Type: application/sdp
Content-Length: 341
v=0
o=SIP 1439163165 1439163294 IN IP4 192.168.179.59
s=Phone-Call
c=IN IP4 192.168.179.59
t=0 0
m=audio 11796 RTP/AVP 0 8 18 9 101 <tel:08189101>
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:30
a=sendrecv
Best Regards,
Mert METİN
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