[OpenSIPS-Users] OpenSIPS as Teams SBC
johan
johan at democon.be
Mon May 11 17:32:42 EST 2020
Ovidiu
do you have any idea why refer-to user part can be empty in refer coming
from teams ?
On 11/05/2020 19:14, Ovidiu Sas wrote:
> Microsoft’s SIP routing is RFC compliment.
> 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.
> 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.
>
> Regards,
> Ovidiu Sas
>
> On Mon, May 11, 2020 at 05:51 Slava Bendersky via Users
> <users at lists.opensips.org <mailto:users at lists.opensips.org>> wrote:
>
> Hello All,
> 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.
> Opensips is rely on route headers and use different way to route it.
>
> volga629
>
> ------------------------------------------------------------------------
> *From: *"John Quick" <john.quick at smartvox.co.uk
> <mailto:john.quick at smartvox.co.uk>>
> *To: *"OpenSIPS users mailling list" <users at lists.opensips.org
> <mailto:users at lists.opensips.org>>, james at ip-sentinel.com
> <mailto:james at ip-sentinel.com>
> *Sent: *Monday, May 11, 2020 6:19:50 AM
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS as Teams SBC
>
> I agree completely with Ovidiu.
> The Microsoft documentation says to use a FQDN in the Contact
> header, but
> this is wrong when the SBC is acting as a SIP Proxy.
> The blog post on the OpenSIPS website explains that actually the
> Record-Route header needs the FQDN.
> The one exception to this is the handling of OPTIONS pings - for
> these,
> OpenSIPS is the end point so it must use a FQDN in Contact.
>
> If you change the Contact header in call setup then you risk
> breaking the
> path for sequential requests, such as ACK.
> If ACK does not reach its destination, the call drops at one end
> after about
> 20 seconds - exactly what you are seeing.
>
> I have not yet found a good way to capture TLS encoded SIP. In
> theory, you
> can use sngrep with the -k option to identify the path to the
> private key
> file.
> It would be necessary to start sngrep first, then start (or restart)
> OpenSIPS. However, this never works for me.
> I had more success using the siptrace module to capture the
> packets to a DB
> table. Presenting it as a sequence diagram may be possible using the
> OpenSIPS Control Panel.
> Wireshark also has the ability to capture, decode and present TLS
> encrypted
> SIP.
> Another option might be to mirror the traffic to Homer in HEP
> format and
> then use Homer to create the sequence diagram.
>
> John Quick
> Smartvox Limited
>
>
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> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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