Hi John,
Thanks for the references. I am still struggling with this though. The
response I am getting from Teams is either SdpParsingError or
BackToBackSessionTerminated and I cannot determine why. Do you happen to have
a working example?
Thanks!
Mark
> On Apr 18, 2020, at 9:26 AM, John Qu
What is freaky is if I just remove the Contact header from the invite entirely
I get a response from Teams.
So now I have Ringing working from Freeswitch to MSTeams. However the CLI is
screwed up and showing a random number that isn’t formatted at all - 266696687
I can answer in Teams but that
Most likely the AudioCodec, Oracle and Ribbon are B2BUAs and there is no
Route/Record-Route routing.
Read about loose routing mechanism in SIP protocol to get a better
understanding on how this headers are used in routing.
Regards,
Ovidiu Sas
On Tue, May 12, 2020 at 10:55 James Hogbin wrote:
>
So I’ve now restarted my script from scratch.
I’ve distilled what I’ve understood from everybody’s suggestions. BUT There is
a total disconnect between the advice and what I can find elsewhere
The AudioCodec, Oracle and Ribbon SBC instructions all say you have to modify
the Contact and don’t me
OK I fixed the RTPProxy to request a port in the correct range of teams media &
I get a response but it’s now I get a rejection.
It’s probably a basic error but I’m pulling my hair out on this one.
James Hogbin
Director
IP Sentinel
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-se
OK. On the Freeswitch side there is a double transport=tls but I can’t find
out where that’s happening so I’ll remove it in the Opensips script
Here’s what Opensips sees from Freeswitch
James Hogbin
Director
IP Sentinel
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com
vidiu 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 p
uration 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 Use
Ovidiu is right,
MS follows RFC. Nothing special, no magic.
Yes, blog article assumes some basic understanding of how SIP proxy works,
but all the necessary steps are there. And of course, there is no single
universal solution for everybody.
You don't need to touch To, From, VIA headers and espe
d all is good.
>
> Regards,
> Ovidiu Sas
>
> On Mon, May 11, 2020 at 05:51 Slava Bendersky via Users
> wrote:
>>
>> Hello All,
>> Microsoft is rely on approved sbc vendors, where most sbc are use VIA and
>> headers to route traffic. That why Contact
and to.
>> Opensips is rely on route headers and use different way to route it.
>>
>> volga629
>>
>>
>> From: "John Quick"
>> To: "OpenSIPS users mailling list" ,
>> ja...@ip-sentinel.com
>> Sent: Monday, May 11, 2020
From: "John Quick" mailto:john.qu...@smartvox.co.uk>>
To: "OpenSIPS users mailling list"
mailto:users@lists.opensips.org>>,
ja...@ip-sentinel.com<mailto:ja...@ip-sentinel.com>
Sent: Monday, May 11, 2020 6:19:50 AM
Subject: Re: [OpenSIPS-Users] OpenSIPS as Teams SBC
.@ip-sentinel.com
<mailto:ja...@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 th
penSIPS users mailling list" ,
> ja...@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
"OpenSIPS users mailling list" ,
ja...@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 a
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 handli
The MS documentation should be taken with a grain of salt. For SIP proxies
there’s no need for an FQDN in the Contact header, instead an FQDN is
required in the top Record-Route header. The FQDN is required in the
Contact header of the OPTIONS ping because this is a direct message between
the proxy
OK.
Thank you for the Record_Route_Preset fix. I had followed the example without
understanding what it was doing. It always seemed odd to put the 5060 port
after the TLS port. I understand why now. And more importantly why not to do
it.
Adding a * to the listen port caused an error in the
James,
Just to add to Alexey's last comment, if your SBC *is* changing the
transport from UDP to TLS, then you must ensure the correct socket is used
on the SBC when it is sending requests to the Teams Proxy:
force_send_socket(tls:137.117.136.143:5091);
You may also find it helpful to add " AS sb
Hi James,
According to your Record-Route headers, OpenSIPS is changing transport from
TLS to UDP for replies to your INVITE.
But in Contact there is transport TLS.
So how is FreeSWITCH connected to OpenSIPS over UDP or TLS?
And if it's TLS, so your rr should look like this:
record_route_preset("sb
That fixed the hangup issue. Thank you very much. I’d’ve never spotted that
in a million years of looking.
Odd that the audio worked at all though
I’m still confused as to why the Teams can route out but the pbx cannot route
back. Other than adding the Record route I’m not doing anything dif
Ok James,
Can you please look at your config with more attention:
if(has_body("application/sdp")){
xlog("[RTPPROXY] route[relay] we have sdp on this message\n$rm\n");
rtpproxy_offer("co", "137.117,136.143");
}
}
xlog("[INFO] Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");
if (!t_relay()) {
send_rep
That is not a typo
No clue as to how to change that. The RTPProxy is set up as per the
documentation. Well as far as I understand it.
I don’t understand how it works fine from Teams to The PBX
But not the other way.
Also how the return path for BYE isn’t working
James Hogbin
Director
IP
Indeed.
On Thu, 7 May 2020, 20:15 Alexey Vasilyev,
wrote:
> Hi James,
>
> Just to be sure, that this is not a typo:
>
> check your SDP
> o= 137.117,136.143
> c=IN IP4 137.117,136.143
>
> This will definitely fail.
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
> http://opensips-ope
Hi James,
Just to be sure, that this is not a typo:
check your SDP
o= 137.117,136.143
c=IN IP4 137.117,136.143
This will definitely fail.
-
---
Alexey Vasilyev
--
Sent from:
http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
_
Your contact header looks wrong. Hint: look in the oracle sbc ms teams docs.
On Thu, 7 May 2020, 18:24 James Hogbin, wrote:
> My set up
>
> Teams <—> sbc.ip-sentinel.com <—> pbx.ip-sentinel.com
>
> I can create a call with audio from Teams -> PBX
> I can hang up that call from Teams but not the
My set up
Teams <—> sbc.ip-sentinel.com <—> pbx.ip-sentinel.com
I can create a call with audio from Teams -> PBX
I can hang up that call from Teams but not the PBX
I cannot route a call from PBX to teams. Although the initial TLS handshake
does happen
I’m pretty sure it’s something basic to do
I have written a couple of articles which, between them, should help you
with this question.
The first article looks at WebRTC <--> SIP using rtpengine:
https://kb.smartvox.co.uk/opensips/webrtc-using-opensips-and-rtpengine/
The other one discusses how you configure OpenSIPS 2.2.x for TLS:
https://
I have configured OpenSIPS to act as an SBC for MS Teams. My voip provider
(voip.ms) supports TLS, so my outgoing calls from MS Teams work great. On the
incoming side though, voip.ms does not support TLS 1.2 so I’m just using TCP or
UDP unsecured. Of course when the call goes out to the MS Te
Hi all
I was asked today about the possibility of using OpenSIPS as an SBC for
Teams/Direct Routing.
I've seen the how to:
https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/
I am just wondering if anyone on this list does this in production? If so
what is your approach to support give
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