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<p>Ovidiu <br>
</p>
<p>do you have any idea why refer-to user part can be empty in refer
coming from teams ?<br>
</p>
<div class="moz-cite-prefix">On 11/05/2020 19:14, Ovidiu Sas wrote:<br>
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<div dir="auto">Microsoft’s SIP routing is RFC compliment.</div>
</div>
<div dir="auto">There’s no special routing for approved SBCs. The
routing Is based on the type of SBC: B2BUA vs proxy, which
again, is rfc complient.</div>
<div dir="auto">For OpenSiPS, which is a proxy, all the
configuration steps are very well outlined in the blog. No need
to mess with Via or Contact headers! Follow the loose routing
rules as outlined in the rfc and all is good.</div>
<div dir="auto"><br>
</div>
<div dir="auto">Regards,</div>
<div dir="auto">Ovidiu Sas</div>
<div><br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Mon, May 11, 2020 at
05:51 Slava Bendersky via Users <<a
href="mailto:users@lists.opensips.org"
moz-do-not-send="true">users@lists.opensips.org</a>>
wrote:<br>
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<div>Hello All,</div>
<div>Microsoft is rely on approved sbc vendors, where
most sbc are use VIA and headers to route traffic.
That why Contact header is important, also they use
from and to.</div>
<div>Opensips is rely on route headers and use different
way to route it.</div>
<div><br>
</div>
<div>volga629 </div>
<div><br>
</div>
<hr id="m_-2318876661958670388zwchr">
<div><b>From: </b>"John Quick" <<a
href="mailto:john.quick@smartvox.co.uk"
target="_blank" moz-do-not-send="true">john.quick@smartvox.co.uk</a>><br>
<b>To: </b>"OpenSIPS users mailling list" <<a
href="mailto:users@lists.opensips.org"
target="_blank" moz-do-not-send="true">users@lists.opensips.org</a>>,
<a href="mailto:james@ip-sentinel.com" target="_blank"
moz-do-not-send="true">james@ip-sentinel.com</a><br>
<b>Sent: </b>Monday, May 11, 2020 6:19:50 AM<br>
<b>Subject: </b>Re: [OpenSIPS-Users] OpenSIPS as
Teams SBC<br>
</div>
</div>
</div>
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<div style="font-family:lucida
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<div><br>
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<div>I agree completely with Ovidiu.<br>
The Microsoft documentation says to use a FQDN in the
Contact header, but<br>
this is wrong when the SBC is acting as a SIP Proxy.<br>
The blog post on the OpenSIPS website explains that
actually the<br>
Record-Route header needs the FQDN.<br>
The one exception to this is the handling of OPTIONS
pings - for these,<br>
OpenSIPS is the end point so it must use a FQDN in
Contact.<br>
<br>
If you change the Contact header in call setup then
you risk breaking the<br>
path for sequential requests, such as ACK.<br>
If ACK does not reach its destination, the call drops
at one end after about<br>
20 seconds - exactly what you are seeing.<br>
<br>
I have not yet found a good way to capture TLS encoded
SIP. In theory, you<br>
can use sngrep with the -k option to identify the path
to the private key<br>
file.<br>
It would be necessary to start sngrep first, then
start (or restart)<br>
OpenSIPS. However, this never works for me.<br>
I had more success using the siptrace module to
capture the packets to a DB<br>
table. Presenting it as a sequence diagram may be
possible using the<br>
OpenSIPS Control Panel.<br>
Wireshark also has the ability to capture, decode and
present TLS encrypted<br>
SIP.<br>
Another option might be to mirror the traffic to Homer
in HEP format and<br>
then use Homer to create the sequence diagram.<br>
<br>
John Quick<br>
Smartvox Limited<br>
<br>
<br>
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-- <br>
<div dir="ltr" class="gmail_signature"
data-smartmail="gmail_signature">VoIP Embedded, Inc.<br>
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