Hi Denys and A Happy New Year,
Let me check the pcap you PM'ed me.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
On 12/22/21 10:18 AM, Denys Pozniak wrote:
Hello!
Yes, that's right, the documentation did not indicate that TH should
generate different Call-IDs for different incoming branches...
But now there is still an open question about the work of the B2B
module. It just generates separate Call-IDs, but does not forward the
SIP CANCEL message (I will share the trace in a private message).
Happy upcoming holidays!
вт, 21 дек. 2021 г. в 17:28, Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>:
Hi Denys,
Doing TH with dialog does not provide you with different call-ids
for each branch. The TH (or changing) is done between in (caller)
and out (callee) sides. There is no doc stating that each branch
will get a different Call-ID (I hope :D).
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com <https://www.opensips-solutions.com>
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
<https://opensips.org/training/OpenSIPS_eBootcamp_2021/>
On 12/14/21 2:13 PM, Denys Pozniak wrote:
Hello!
Bogdan,
I tested the combination of dialog + TH modules and found out
that this also does not work correctly if the incoming call was
forked.
Outgoing legs have the same Call-ID and tag, although I would
expect them to be different.
The configuration is exactly the same as in the
Documentation/Tutorials-Topology-Hiding
[root@f-proxy opensips]$ opensips -V
version: opensips 3.2.3 (x86_64/linux)
ср, 6 окт. 2021 г. в 12:18, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>>:
Hi Denys,
Before diving into the B2B dark corners, I would strongly
suggest to use OpenSIPS with dialog + topology hiding
modules, rather than B2B. The B2B is not so friendly with
parallel forking.
And as time as you only need TH, dialog + TH is be best way
to do it.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
<https://www.opensips-solutions.com>
OpenSIPS eBootcamp 2021
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
<https://opensips.org/training/OpenSIPS_eBootcamp_2021/>
On 9/7/21 2:14 PM, Denys Pozniak wrote:
Adding a scheme of the current call-flow scenario maybe it
is not completely clear from the previous message:
FreeSWITCH --(1-call)--> Fork Proxy --(N-branches)--> SEMS
--(N-calls)--> Edge Proxy ----> N-devices
вт, 7 сент. 2021 г. в 12:57, Denys Pozniak
<[email protected] <mailto:[email protected]>>:
Hello!
Our service delivery logic is as follows:
Each user has an internal extension, under which there
are several devices with their own identifier.
With an incoming call to such a subscriber, FreeSWITCH
adds custom SIP headers with these device identifiers.
Further on these fields the superior Proxy forks legs
and then these legs pass through the Sems to become
completely separate calls.
Now there is a task to replace Sems with OpenSIPS.
The script below works fine, but only if the incoming
calls are not forked
/####### Routing Logic ########
route{
if (is_method("INVITE") && !has_totag()) {
b2b_init_request("top hiding");
exit;
}
}
route[b2b_logic_request] {
b2b_pass_request();
exit;
}/
If there is a fork with an answer on some device, then
OpenSIPS does not forward the SIP CANCEL (Reason:
SIP;cause=200;text="Call completed elsewhere") to the
rest and these devices keep ringing until timeout
(Reason: SIP;cause=480;text="NO_ANSWER")
Please help understand the nature of this behavior.
version: opensips 3.2.2 (x86_64/linux)
*Incoming SIP INVITE:*
2021/09/07 11:38:30.737456 192.168.27.84:5060
<http://192.168.27.84:5060> -> 192.168.27.84:5080
<http://192.168.27.84:5080>
INVITE
sip:[email protected]:5060;transport=udp SIP/2.0
Record-Route:
<sip:192.168.27.84;lr=on;ftag=3a8gNpgZQ89pj;did=8b.4a4;vst=AAAAAEcYQ0JfBhUaEEoOFQAAAAAAAAAAAAAJBjY->
Record-Route: <sip:192.168.27.126;lr=on;did=8b01.a1d4>
Via: SIP/2.0/UDP
192.168.27.84;branch=z9hG4bKcc18.ec9a363ccc70d07691e11293d160cca6.1
Via: SIP/2.0/UDP
192.168.27.126;branch=z9hG4bKcc18.accd8d8bac35ac66a172f6ce173c9a34.0
Via: SIP/2.0/UDP
192.168.27.123;received=192.168.27.123;rport=5060;branch=z9hG4bKavcjKF58g9D1e
Max-Forwards: 66
From: "VOIP" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=3a8gNpgZQ89pj
To: <sip:[email protected]
<mailto:sip%[email protected]>>
Call-ID: 33e8140a-8a62-123a-e1ba-001dd8b71cb2
CSeq: 40949963 INVITE
Contact: <sip:[email protected]:5060
<http://sip:[email protected]:5060>>
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
*Outgoing SIP INVITE:*
2021/09/07 11:38:30.737938 192.168.27.84:5080
<http://192.168.27.84:5080> -> 192.168.27.126:5060
<http://192.168.27.126:5060>
INVITE
sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
192.168.27.84:5080;branch=z9hG4bK6ddf.d88b07f2.0
To: sip:[email protected]:5060
<http://sip:[email protected]:5060>
From: "VOIP" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=94fd20254e546fee730f360cf9860800
CSeq: 40949964 INVITE
Call-ID: B2B.331.6374211.1631007510
Max-Forwards: 70
Content-Length: 486
User-Agent: OpenSIPS (3.2.2 (x86_64/linux))
Content-Type: application/sdp
Supported: timer, path, replaces
P-Asserted-Identity: " VOIP" <sip:[email protected]
<mailto:sip%[email protected]>>
Privacy: none
Content-Disposition: session
X-Call-ID: 33e8140a-8a62-123a-e1ba-001dd8b71cb2
Contact: <sip:[email protected]:5080
<http://sip:[email protected]:5080>>
*Incoming SIP CANCEL:*
2021/09/07 11:38:33.593381 192.168.27.84:5060
<http://192.168.27.84:5060> -> 192.168.27.84:5080
<http://192.168.27.84:5080>
CANCEL
sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
192.168.27.84;branch=z9hG4bKcc18.ec9a363ccc70d07691e11293d160cca6.1
Max-Forwards: 66
From: "VOIP" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=3a8gNpgZQ89pj
To: <sip:[email protected]
<mailto:sip%[email protected]>>
Call-ID: 33e8140a-8a62-123a-e1ba-001dd8b71cb2
CSeq: 40949963 CANCEL
Content-Length: 0
Reason: SIP;cause=200;text="Call completed elsewhere"
*Outgoing SIP CANCEL by timeout (with 27 sec delay):*
2021/09/07 11:39:01.100888 192.168.27.84:5080
<http://192.168.27.84:5080> -> 192.168.27.126:5060
<http://192.168.27.126:5060>
CANCEL
sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP
192.168.27.84:5080;branch=z9hG4bK6ddf.d88b07f2.0
From: "VOIP" <sip:[email protected]
<mailto:sip%[email protected]>>;tag=94fd20254e546fee730f360cf9860800
Call-ID: B2B.331.6374211.1631007510
To: sip:[email protected]:5060
<http://sip:[email protected]:5060>
CSeq: 40949964 CANCEL
Max-Forwards: 70
Reason: SIP;cause=480;text="NO_ANSWER"
User-Agent: OpenSIPS (3.2.2 (x86_64/linux))
Content-Length: 0
--
BR,
Denys Pozniak
--
BR,
Denys Pozniak
_______________________________________________
Users mailing list
[email protected] <mailto:[email protected]>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
--
BR,
Denys Pozniak
--
BR,
Denys Pozniak
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
