Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.7, 2.3.4 and 2.4.1 minor releases

[Esty, Ryan]

Răzvan,

This is the in your tracking software 
https://github.com/OpenSIPS/opensips/issues/1364.

Ryan Esty
Senior Software Engineer
NEC Enterprise Communication Technologies (Cheshire)
203-718-6268


-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Thursday, May 17, 2018 9:00 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.7, 2.3.4 and 2.4.1 minor 
releases

Hi, Ryan!

I think you are right, rtcp-mux-require might not be accepted by the current 
code of rtpengine. Please open an issue on our ticketing system [1] to keep 
track of this issue as well.

[1] https://github.com/OpenSIPS/opensips/issues

Best regards,
Răzvan

On 05/17/2018 03:27 PM, Esty, Ryan wrote:
> Bogdan-Andrei,
> 
> I was going to look into this a little more before reporting it but I'm 
> having issues trying to get rtcp-mux-require to be sent down to rtpengine. 
> From what I can tell I need this to work with chrome's webrtc. Since this is 
> crunch time I didn't want a potential bug not being covered. I'm using 2.3.3 
> and it could be that rtcp-mux-require isn't part of that version. From what I 
> can tell so far if I put in rtcp-mux-require rtpengine kicks it back. When I 
> do a rtcp-mux=require rtpengine doesn't complain but the latter doesn't match 
> the documentation nor do I think works.
> 
> Ryan Esty
> 
> -Original Message-
> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of 
> Bogdan-Andrei Iancu
> Sent: Thursday, May 17, 2018 6:40 AM
> To: users@lists.opensips.org; developensips <de...@lists.opensips.org>
> Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.7, 2.3.4 and 2.4.1 
> minor releases
> 
> Greetings!
> 
> A new cycle is about to be complete: OpenSIPS is tortured all over the 
> worlds; this produces feedback and reports; and this translates into more 
> fix, in a better stability of the released OpenSIPS versions.
> 
> Shortly, it's time for a new set of minor releases, to officially 
> incorporate all the fixes in the last months/weeks: OpenSIPS 2.2.7,
> 2.3.4 and 2.4.1
> 
> The new releases are due next week, on Wednesday, May 24th 2018.
> 
> If you have any pending GitHub issues/mailing list bug threads concerning the 
> mentioned branches, this would be a good time to bump them!
> 
> Thank you for your contributions to this project!
> 
> Best regards,
> 
> --
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> OpenSIPS Summit 2018
> http://www.opensips.org/events/Summit-2018Amsterdam
> 
> 
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

--
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com

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Re: [OpenSIPS-Users] from address in the t_relay

[Esty, Ryan]

List,

I was able to get around my problem by disabling tcp_async. I was seeing a 
whole bunch of Polling is overdue. This doesn't seem like the right solution 
though, more like I fixed it by some timing side effect.

Ryan

From: Esty, Ryan
Sent: Wednesday, April 18, 2018 8:59 AM
To: 'Users@lists.opensips.org' <Users@lists.opensips.org>
Subject: from address in the t_relay

Hi list,

I'm hoping someone can direct me to some documentation or help me out. I'm 
trying to send an invite to something outside my domain. Everything seems to 
work but no packets come back to opensips. I see the packets on the wire coming 
to the machine but it looks like the tcp port that was used to send stuff out 
isn't there anymore. A colleage found this gem in the log: "proto_tcp_send: 
Successfully connected from interface 209.197.207.36:5060 to 
209.197.207.36:5060" thinking that maybe the tcp connection was opened by 
mistake with the local side not set correctly.

My question is when t_relay is called how does it know the from address? I 
printed out my from domain $fd and it is using my internal domain, which works 
out to be a 192.168 address. The address listed above is from a successfully 
DNS look up on the request URI domain and it works out to be the above address.

Ryan

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[OpenSIPS-Users] from address in the t_relay

[Esty, Ryan]

Hi list,

I'm hoping someone can direct me to some documentation or help me out. I'm 
trying to send an invite to something outside my domain. Everything seems to 
work but no packets come back to opensips. I see the packets on the wire coming 
to the machine but it looks like the tcp port that was used to send stuff out 
isn't there anymore. A colleage found this gem in the log: "proto_tcp_send: 
Successfully connected from interface 209.197.207.36:5060 to 
209.197.207.36:5060" thinking that maybe the tcp connection was opened by 
mistake with the local side not set correctly.

My question is when t_relay is called how does it know the from address? I 
printed out my from domain $fd and it is using my internal domain, which works 
out to be a 192.168 address. The address listed above is from a successfully 
DNS look up on the request URI domain and it works out to be the above address.

Ryan

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Re: [OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Razvan,

Thanks for the info I'll get the latest 2.3 and compile it.

Ryan Esty
Senior Software Engineer
NEC Enterprise Communication Technologies, Inc.  (Cheshire)
203-718-6268


From: Users <users-boun...@lists.opensips.org> on behalf of Răzvan Crainea 
<raz...@opensips.org>
Sent: Monday, April 16, 2018 2:41 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi, Ryan!

That error is actually triggered by opensips, rtpengine module. There
was a bug related to this issue that was fixed on 21st of February, and
did not catch the 2.3.3 release. The latest 2.3 nightly has this fixed,
so I'd suggest you to use that opensips version.

Best regards,
Răzvan

On 04/13/2018 04:31 PM, Esty, Ryan wrote:
> Razvan,
>
> Rtpengine is printing out this: rtpengine:parse_flags: error processing flag 
> `codec-strip-VP8': unknown error. As I look at it now I don't see how it 
> could be your issue. You aren't modifying the flag I sent to rtpengine. If it 
> helps my version of opensips is 2.3.3.
>
> Ryan
>
>
> -Original Message-
> From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
> Crainea
> Sent: Friday, April 13, 2018 9:17 AM
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine
>
> Hi, Ryan!
>
> I think the issue you are talking about is related to OpenSIPS, rather than 
> rtpengine, since you are getting the error in OpenSIPS, is that right?
>
> Can you confirm what version of OpenSIPS you are using?
>
> Best regards,
> Răzvan
>
> On 04/13/2018 03:42 PM, Esty, Ryan wrote:
>> Bogdan-Andrei,
>>
>> Thanks for the information just in case someone else looks for this,
>> this is the tracker https://github.com/sipwise/rtpengine/issues/525.
>>
>> Ryan
>>
>> *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
>> *Sent:* Thursday, April 12, 2018 4:08 PM
>> *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty,
>> Ryan <ryan.e...@necect.com>
>> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine
>>
>> Hi Ryan,
>>
>> yeah, this happens because OpenSIPS applies all the changes at the
>> end, when the message is about to be sent out. As a side effect, when
>> sending the SDP to rtpengine, opensips does not see its own previous
>> changes over the body (changes are still pending).
>> Usually there are easy workarounds for this, but in this case it looks
>> like bug to me. Could you please open a bug report the the github tracker.
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>
>> http://www.opensips-solutions.com
>>
>> OpenSIPS Summit 2018
>>
>> http://www.opensips.org/events/Summit-2018Amsterdam
>>
>> On 04/09/2018 05:22 PM, Esty, Ryan wrote:
>>
>>  Hi opensips list,
>>
>>  First some background I’m trying to use opensips as a webrtc proxy.
>>  I found out that the packets for the invite going to my sip server
>>  are too big for my sip server. It doesn’t like packets to be over
>>  4000 bytes. I’m trying to take what I can out of the sip packets
>>  like codes I know the other side can’t do. First codec stripping
>>  works but only with the audio codecs. If I try to strip a video
>>  codec the packet gets mangled. This is probably a bug in rtpengine
>>  and not opensips. I was hoping if anyone has any idea how I might
>>  get my invite packets smaller? The webrtc side is generating ssrc
>>  lines in my sdp. I’m trying to strip them out but I’m not sure if
>>  rtpengine is putting them back in or not. Before my rtpengine_offer
>>  I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has
>>  all the ssrc lines in it.
>>
>>  Ryan
>>
>>
>>
>>
>>  ___
>>
>>  Users mailing list
>>
>>  Users@lists.opensips.org <mailto:Users@lists.opensips.org>
>>
>>  http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
> --
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
> OpenSIPS Summit 2018
> http://www.opensips.org/events/Summit-2018Amsterdam
>
> __

Re: [OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Razvan,

Rtpengine is printing out this: rtpengine:parse_flags: error processing flag 
`codec-strip-VP8': unknown error. As I look at it now I don't see how it could 
be your issue. You aren't modifying the flag I sent to rtpengine. If it helps 
my version of opensips is 2.3.3.

Ryan


-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Friday, April 13, 2018 9:17 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi, Ryan!

I think the issue you are talking about is related to OpenSIPS, rather than 
rtpengine, since you are getting the error in OpenSIPS, is that right?

Can you confirm what version of OpenSIPS you are using?

Best regards,
Răzvan

On 04/13/2018 03:42 PM, Esty, Ryan wrote:
> Bogdan-Andrei,
> 
> Thanks for the information just in case someone else looks for this, 
> this is the tracker https://github.com/sipwise/rtpengine/issues/525.
> 
> Ryan
> 
> *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> *Sent:* Thursday, April 12, 2018 4:08 PM
> *To:* OpenSIPS users mailling list <users@lists.opensips.org>; Esty, 
> Ryan <ryan.e...@necect.com>
> *Subject:* Re: [OpenSIPS-Users] codec stripping with rtpengine
> 
> Hi Ryan,
> 
> yeah, this happens because OpenSIPS applies all the changes at the 
> end, when the message is about to be sent out. As a side effect, when 
> sending the SDP to rtpengine, opensips does not see its own previous 
> changes over the body (changes are still pending).
> Usually there are easy workarounds for this, but in this case it looks 
> like bug to me. Could you please open a bug report the the github tracker.
> 
> Best regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
> 
>http://www.opensips-solutions.com
> 
> OpenSIPS Summit 2018
> 
>http://www.opensips.org/events/Summit-2018Amsterdam
> 
> On 04/09/2018 05:22 PM, Esty, Ryan wrote:
> 
> Hi opensips list,
> 
> First some background I’m trying to use opensips as a webrtc proxy.
> I found out that the packets for the invite going to my sip server
> are too big for my sip server. It doesn’t like packets to be over
> 4000 bytes. I’m trying to take what I can out of the sip packets
> like codes I know the other side can’t do. First codec stripping
> works but only with the audio codecs. If I try to strip a video
> codec the packet gets mangled. This is probably a bug in rtpengine
> and not opensips. I was hoping if anyone has any idea how I might
> get my invite packets smaller? The webrtc side is generating ssrc
> lines in my sdp. I’m trying to strip them out but I’m not sure if
> rtpengine is putting them back in or not. Before my rtpengine_offer
> I do a replace_body_all(“a=ssrc.*\r\n,” “”) but the invite still has
> all the ssrc lines in it.
> 
> Ryan
> 
> 
> 
> 
> ___
> 
> Users mailing list
> 
> Users@lists.opensips.org <mailto:Users@lists.opensips.org>
> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

--
Răzvan Crainea
OpenSIPS Core Developer
   http://www.opensips-solutions.com
OpenSIPS Summit 2018
   http://www.opensips.org/events/Summit-2018Amsterdam

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Re: [OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Bogdan-Andrei,

Thanks for the information just in case someone else looks for this, this is 
the tracker https://github.com/sipwise/rtpengine/issues/525.

Ryan

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Thursday, April 12, 2018 4:08 PM
To: OpenSIPS users mailling list <users@lists.opensips.org>; Esty, Ryan 
<ryan.e...@necect.com>
Subject: Re: [OpenSIPS-Users] codec stripping with rtpengine

Hi Ryan,

yeah, this happens because OpenSIPS applies all the changes at the end, when 
the message is about to be sent out. As a side effect, when sending the SDP to 
rtpengine, opensips does not see its own previous changes over the body 
(changes are still pending).
Usually there are easy workarounds for this, but in this case it looks like bug 
to me. Could you please open a bug report the the github tracker.

Best regards,


Bogdan-Andrei Iancu



OpenSIPS Founder and Developer

  http://www.opensips-solutions.com

OpenSIPS Summit 2018

  http://www.opensips.org/events/Summit-2018Amsterdam
On 04/09/2018 05:22 PM, Esty, Ryan wrote:
Hi opensips list,

First some background I'm trying to use opensips as a webrtc proxy. I found out 
that the packets for the invite going to my sip server are too big for my sip 
server. It doesn't like packets to be over 4000 bytes. I'm trying to take what 
I can out of the sip packets like codes I know the other side can't do. First 
codec stripping works but only with the audio codecs. If I try to strip a video 
codec the packet gets mangled. This is probably a bug in rtpengine and not 
opensips. I was hoping if anyone has any idea how I might get my invite packets 
smaller? The webrtc side is generating ssrc lines in my sdp. I'm trying to 
strip them out but I'm not sure if rtpengine is putting them back in or not. 
Before my rtpengine_offer I do a replace_body_all("a=ssrc.*\r\n," "") but the 
invite still has all the ssrc lines in it.

Ryan





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[OpenSIPS-Users] codec stripping with rtpengine

[Esty, Ryan]

Hi opensips list,

First some background I'm trying to use opensips as a webrtc proxy. I found out 
that the packets for the invite going to my sip server are too big for my sip 
server. It doesn't like packets to be over 4000 bytes. I'm trying to take what 
I can out of the sip packets like codes I know the other side can't do. First 
codec stripping works but only with the audio codecs. If I try to strip a video 
codec the packet gets mangled. This is probably a bug in rtpengine and not 
opensips. I was hoping if anyone has any idea how I might get my invite packets 
smaller? The webrtc side is generating ssrc lines in my sdp. I'm trying to 
strip them out but I'm not sure if rtpengine is putting them back in or not. 
Before my rtpengine_offer I do a replace_body_all("a=ssrc.*\r\n," "") but the 
invite still has all the ssrc lines in it.

Ryan

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Re: [OpenSIPS-Users] h264 webrtc and opensips

[Esty, Ryan]

Razvan,

So I'm not sure if this helps or not but we just tested Janus and I was able to 
connect our device using that. I must have something wrong with my webrtc 
gateway approach in rtpengine. We would like to use opensips as it is more 
versatile at least we can prove not it isn't our devices. The 
packetization-mode was being set to one in the Janus example also.

Ryan Esty

-Original Message-
From: Esty, Ryan 
Sent: Tuesday, April 3, 2018 1:28 PM
To: users@lists.opensips.org
Subject: RE: [OpenSIPS-Users] h264 webrtc and opensips

Razvan,

Thanks for getting back to me. I was afraid of this as I didn't see any options 
in rtpengine that supported video codecs either. We are trying to upgrade some 
of our devices to support packetization-mode 0 and 1.

Ryan Esty
Senior Software Engineer
NEC Enterprise Communication Technologies (Cheshire)
203-718-6268

-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Tuesday, April 3, 2018 1:19 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] h264 webrtc and opensips

Hi, Ryan!

I don't have that much experience with H.264, but my first instinct was to look 
into the rtpengine packetization feature. But unfortunately rtpengine does not 
support H.264 codecs, so I doubt this can help. But perhaps you could look into 
different transcoding solutions that do support H.264 transcoding.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/23/2018 05:01 PM, Esty, Ryan wrote:
> Hi list,
> 
> This might not be the correct list for this but maybe someone might be 
> able to point me in the correct direction. I’m trying to use opensips 
> as a webrtc gateway. It mostly works I’m able to call a legacy sip 
> phone connected to my SIP server. The reason why it only mostly works 
> is I have a problem with the h264 codec. None of my legacy devices 
> know what to do with packetization-mode=1, well this is my assumption.
> Has anyone else had a similar issue and can point me to some further 
> information? A lot of people said to just set packetization-mode to 0 
> but I thought the webrtc video draft said this was mandatory 
> (https://tools.ietf.org/html/rfc7742).
> 
> Ryan Esty
> 
> 
> 
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

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Re: [OpenSIPS-Users] h264 webrtc and opensips

[Esty, Ryan]

Razvan,

Thanks for getting back to me. I was afraid of this as I didn't see any options 
in rtpengine that supported video codecs either. We are trying to upgrade some 
of our devices to support packetization-mode 0 and 1.

Ryan Esty
Senior Software Engineer
NEC Enterprise Communication Technologies (Cheshire)
203-718-6268

-Original Message-
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Razvan 
Crainea
Sent: Tuesday, April 3, 2018 1:19 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] h264 webrtc and opensips

Hi, Ryan!

I don't have that much experience with H.264, but my first instinct was to look 
into the rtpengine packetization feature. But unfortunately rtpengine does not 
support H.264 codecs, so I doubt this can help. But perhaps you could look into 
different transcoding solutions that do support H.264 transcoding.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/23/2018 05:01 PM, Esty, Ryan wrote:
> Hi list,
> 
> This might not be the correct list for this but maybe someone might be 
> able to point me in the correct direction. I’m trying to use opensips 
> as a webrtc gateway. It mostly works I’m able to call a legacy sip 
> phone connected to my SIP server. The reason why it only mostly works 
> is I have a problem with the h264 codec. None of my legacy devices 
> know what to do with packetization-mode=1, well this is my assumption. 
> Has anyone else had a similar issue and can point me to some further 
> information? A lot of people said to just set packetization-mode to 0 
> but I thought the webrtc video draft said this was mandatory 
> (https://tools.ietf.org/html/rfc7742).
> 
> Ryan Esty
> 
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

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[OpenSIPS-Users] h264 webrtc and opensips

[Esty, Ryan]

Hi list,

This might not be the correct list for this but maybe someone might be able to 
point me in the correct direction. I'm trying to use opensips as a webrtc 
gateway. It mostly works I'm able to call a legacy sip phone connected to my 
SIP server. The reason why it only mostly works is I have a problem with the 
h264 codec. None of my legacy devices know what to do with 
packetization-mode=1, well this is my assumption. Has anyone else had a similar 
issue and can point me to some further information? A lot of people said to 
just set packetization-mode to 0 but I thought the webrtc video draft said this 
was mandatory (https://tools.ietf.org/html/rfc7742).

Ryan Esty

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Re: [OpenSIPS-Users] mid-registrar and WSS

[Esty, Ryan]

All,

Never mind it is registering now. I still have some work to do but after more 
investigation the SIP server was deleting the second VIA which was the via for 
WSS. Now on the register I save it to an avp and then in my register's 
onreply_route I just do an append_hf("Via: ..."). The client sipml5 registers 
and now I can try making calls.

Ryan



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[OpenSIPS-Users] mid-registrar and WSS

[Esty, Ryan]

Hi opensips users,

I have a couple of questions and I'm hoping someone can point me in the right 
direction.

We have a SIP PBX that doesn't do WSS for example using sipml5. I'm trying to 
put opensips in the middle of the SIP PBX and the WSS client with limited 
success using mid-registrar in opensips 2.3. If I don't try to pass the 
registration through and just use opensips as my SIP server the WSS client will 
register and calls can be made. If I do put in the mid-registrar and pass 
through the registration to the SIP PBX the SIP PBX will register the phone and 
pass back the 200 OK the client gets a message back but doesn't think it is for 
him because I'm assuming it is missing the VIA. On a successful registration 
the client will see the VIA on the unsuccessful registration the client doesn't 
see the VIA.


My first question is am I using the mid-registrar as designed I'm trying to set 
it up as contact mirroring (mode 0)? If I did pick the correct method why is my 
VIA disappearing and can I get it back? Someone on the list mentioned this 
https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations and I will 
try it soon but it seemed the mid-registrar which I don't think was released 
yet was supposed to automatically do this.

This is the debug output when the registration is failed:
SEND: REGISTER sip:192.168.40.175 SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKd7RGgr1eBtcOk2yUuIUuyvIqWk3G3Or0;rport
From: "Skwisgaar Skwigelf";tag=T9XcAa1Am5fIDrtIo68Z
To: "Skwisgaar Skwigelf"
Contact: "Skwisgaar 
Skwigelf";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 766574bd-8060-562d-1f67-24975848739c
CSeq: 5334 REGISTER
Content-Length: 0
Route: 
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 ==session event = sent_request
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 200 OK
From: "Skwisgaar Skwigelf";tag=T9XcAa1Am5fIDrtIo68Z
To: "Skwisgaar 
Skwigelf";tag=83a00482-48f7-4f78-ab314b6aacadc545
Contact: 
;expires=200
Call-ID: 766574bd-8060-562d-1f67-24975848739c
CSeq: 5334 REGISTER
Content-Length: 0
Server: Sphericall/9.1.0 Build/669
Allow: 
INVITE,ACK,CANCEL,OPTIONS,REFER,BYE,REGISTER,SUBSCRIBE,NOTIFY,UPDATE,PRACK,MESSAGE,INFO
Allow-Events: message-summary,presence

This is the debug output when the registration is successful:
SEND: REGISTER sip:192.168.40.175 SIP/2.0
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKLpgkzXTPZl5D5ghXuZ1vmNZXUn9nHkMr;rport
From: "Skwisgaar Skwigelf";tag=RyaJ7SLKEeeTS8r9v0BX
To: "Skwisgaar Skwigelf"
Contact: "Skwisgaar 
Skwigelf";expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 6e7fe61e-a51f-96cf-a9a0-e3c3785259b9
CSeq: 7252 REGISTER
Content-Length: 0
Route: 
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;rport=60161;received=172.25.111.53;branch=z9hG4bKLpgkzXTPZl5D5ghXuZ1vmNZXUn9nHkMr
From: "Skwisgaar Skwigelf";tag=RyaJ7SLKEeeTS8r9v0BX
To: "Skwisgaar 
Skwigelf";tag=54638b9941aeeb05912f3f3563e30b77.7a8c
Contact: 
;expires=200;received="sip:172.25.111.53:60161;transport=WSS"
Call-ID: 6e7fe61e-a51f-96cf-a9a0-e3c3785259b9
CSeq: 7252 REGISTER
Content-Length: 0
Server: OpenSIPS (2.3.3 (x86_64/linux))

Ryan

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