Re: [SR-Users] Installing rtpengine from RPM on CentOS

[Pan Christensen]

That’s the kamailio module to control rtpengine, not the rtpengine itself.


Pan B. Christensen
Developer
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

 [mail_footer]

From: sr-users  On Behalf Of Sergey Safarov
Sent: torsdag 1. november 2018 13:37
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Installing rtpengine from RPM on CentOS

This is packaged in main kamailio rpm package
https://github.com/kamailio/kamailio/blob/master/pkg/kamailio/obs/kamailio.spec#L1453-L1454

чт, 1 нояб. 2018 г. в 14:33, Pan Christensen 
mailto:pan.christen...@phonect.no>>:
Hello!

According to https://github.com/sipwise/rtpengine/blob/master/el/README.el.md , 
we should be able to install rtpengine from a CentOS repository using yum. 
We’re unable to find the packages. Any suggestions?


Pan B. Christensen
Developer
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

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[SR-Users] Installing rtpengine from RPM on CentOS

[Pan Christensen]

Hello!

According to https://github.com/sipwise/rtpengine/blob/master/el/README.el.md , 
we should be able to install rtpengine from a CentOS repository using yum. 
We're unable to find the packages. Any suggestions?


Pan B. Christensen
Developer
Phonect AS
Mail: pan.christen...@phonect.no
+ 47 41 88 88 00

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Re: [SR-Users] Missing documentation ims_usrloc_scscf module

[Pan Christensen]

Thanks Henning and Carsten.

It should also be noted that the module is required by both IMS_ISC and 
IMS_REGISTRAR_SCSCF. The lacking documentation of IMS_USRLOC_SCSCF also make 
those modules difficult to implement.

Pan B. Christensen 
Developer
Phonect AS


> -Original Message-
> From: Henning Westerholt 
> Sent: tirsdag 16. oktober 2018 20:27
> To: sr-users@lists.kamailio.org
> Cc: Carsten Bock ; Pan Christensen
> 
> Subject: *** SPAM *** Re: [SR-Users] Missing documentation
> ims_usrloc_scscf module
> 
> Am Dienstag, 16. Oktober 2018, 13:31:08 CEST schrieb Carsten Bock:
> > can you open a ticket on the GitHub Tracker for this? Otherwise this
> > might get lost at some point... I don't have time to take care of it
> > immediately as I am travelling this and next week, more likely I would
> > take care of it later in November.
> 
> Hello Carsten,
> 
> i have opened already a ticket about that one month ago. It is already
> assigned to ngvoice. ;-)
> 
> https://github.com/kamailio/kamailio/issues/1644
> 
> Best regards,
> 
> Henning
> 
> --
> Henning Westerholt - https://skalatan.de/blog/ Kamailio security assessment
> - https://skalatan.de/de/assessment

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Re: [SR-Users] Missing documentation

[Pan Christensen]

The documentation for this module seems to be missing everywhere:

I've checked 4.0, 4.1, 4.2, 4.3, 4.4, 5.0 and 5.1 in addition to stable and 
devel.

Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

 [mail_footer]

From: sr-users  On Behalf Of Pan 
Christensen
Sent: tirsdag 16. oktober 2018 09:43
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Missing documentation

Also missing from devel:

Not Found

The requested URL /docs/modules/devel/modules/ims_usrloc_scscf.html was not 
found on this server.


Apache/2.4.10 (Debian) Server at www.kamailio.org<http://www.kamailio.org> Port 
443


Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

 [mail_footer]

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Pan Christensen
Sent: tirsdag 16. oktober 2018 09:14
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Missing documentation

Hello.

The documentation for the ims_usrloc_scscf module seems to be missing:

Not Found
The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not 
found on this server.

Apache/2.4.10 (Debian) Server at www.kamailio.org<http://www.kamailio.org> Port 
443



Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

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Re: [SR-Users] Missing documentation

[Pan Christensen]

Also missing from devel:

Not Found

The requested URL /docs/modules/devel/modules/ims_usrloc_scscf.html was not 
found on this server.


Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443


Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

 [mail_footer]

From: sr-users  On Behalf Of Pan 
Christensen
Sent: tirsdag 16. oktober 2018 09:14
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Missing documentation

Hello.

The documentation for the ims_usrloc_scscf module seems to be missing:

Not Found
The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not 
found on this server.

Apache/2.4.10 (Debian) Server at www.kamailio.org<http://www.kamailio.org> Port 
443



Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
+ 47 41 88 88 00

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[SR-Users] Missing documentation

[Pan Christensen]

Hello.

The documentation for the ims_usrloc_scscf module seems to be missing:

Not Found
The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not 
found on this server.

Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443



Pan B. Christensen
Utvikler
Phonect AS
Mail: pan.christen...@phonect.no
+ 47 41 88 88 00

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Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[Pan Christensen]

Or maybe FreeSwitch is redundant if you use rtpengine…

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Emanuel 
Gianico
Sent: fredag 15. juni 2018 13:29
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

I'm going to investigate Kazoo samples as Gorlichenko suggested because I think 
using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch 
fully supports WebRTC

El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko 
mailto:ovoshl...@gmail.com>> escribió:
You can watch at the kazoo project examples if you want to avoid rtp proxy

On Thu, Jun 14, 2018, 23:26 Daniel Tryba 
mailto:d.tr...@pocos.nl>> wrote:
On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> From the logs I see the jssip throw this error:
>
> "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
>
> I would like to avoid RTPEngine, because from what I understand, FreeSwitch
> can handle the media part.

IIRC I got the same error in my tries to transcode/bridge SIP over TLS
with SRTP to just plain old SIP with RTP. I haven't put any effort in it
to get it working. You'll need to play around with rtpengine
offer/answer, I based my test on
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
but I blamed my failure on an old rtpengine :)


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Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[Pan Christensen]

> On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote:
> > > I based my
> > > test on https://github.com/havfo/WEBRTC-to-
> > > SIP/blob/master/etc/kamailio/kamailio.cfg
> > > but I blamed my failure on an old rtpengine :)
> >
> > You need to add 'SDES-off' to the rtpengine_manage strings for calls going
> to WebRTC. Most browsers don't support fallback to SDES (anymore) and will
> reject the call if both DTLS and SDES are offered.
> 
> IIRC the problem I had was that going from RTP to SRTP, there was no key
> exchange in SDP (a=crypto) being added by rtpengine. I'll look into this again
> in the near future.

a=crypto is for SDES, and should not be present in SDP. You need a=fingerprint 
for DTLS.
Currently both are added by rtpengine if RTP/SAVPF is specified without SDES-off

With kind regards
Pan B. Christensen
Developer
Phonect AS 


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Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[Pan Christensen]

> I based my
> test on https://github.com/havfo/WEBRTC-to-
> SIP/blob/master/etc/kamailio/kamailio.cfg
> but I blamed my failure on an old rtpengine :)

You need to add 'SDES-off' to the rtpengine_manage strings for calls going to 
WebRTC. Most browsers don't support fallback to SDES (anymore) and will reject 
the call if both DTLS and SDES are offered.


With kind regards
Pan B. Christensen
Developer
Phonect AS 


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Re: [SR-Users] outbound flow tokens and kamailio restart

[Pan Christensen]

Bounce.

Med vennlig hilsen
Pan B. Christensen

From: sr-users  On Behalf Of Pan 
Christensen
Sent: onsdag 13. juni 2018 09:19
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] outbound flow tokens and kamailio restart

Hello all.

I have created a WebRTC to SIP gateway. I implemented it using the outbound 
module. If I restart Kamailio during a call, subsequent messages fail to be 
routed.

{1 322 BYE 218565972_26748892@x.x.x.x<mailto:218565972_26748892@x.x.x.x>} INFO: 
outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validation
{1 322 BYE 218565972_26748892@x.x.x.x<mailto:218565972_26748892@x.x.x.x>} INFO: 
rr [loose.c:519]: process_outbound(): failed to decode flow token
{1 322 BYE 218565972_26748892@x.x.x.x<mailto:218565972_26748892@x.x.x.x>} INFO: 
rr [loose.c:794]: after_loose(): failed to process outbound flow-token

Is it possible to make the flow token survive a restart?

With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
Mobile: 41 88 88 00


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Re: [SR-Users] Asterisk inviting Kamailio

[Pan Christensen]

This question probably belongs in the asterisk forum, but here's a quick answer:

It's most likely a reINVITE in order to modify the existing session. Compare it 
to the original and see if anything has changed (like the codecs in the SDP).

Med vennlig hilsen
Pan B. Christensen
Utvikler
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: onsdag 13. juni 2018 12:47
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Asterisk inviting Kamailio

Hello All,

I have noticed that sometimes when a call is made from one endpoint to another 
through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after 
the call has been established.  Sometimes this does not happen.   When it does 
happen, calls drop.
Why would an INVITE be sent back to Kamailio?

Also, Pan, I am working on the response you requested yesterday.

Thanks you All,
-Steve
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[SR-Users] outbound flow tokens and kamailio restart

[Pan Christensen]

Hello all.

I have created a WebRTC to SIP gateway. I implemented it using the outbound 
module. If I restart Kamailio during a call, subsequent messages fail to be 
routed.

{1 322 BYE 218565972_26748892@x.x.x.x} INFO: outbound [outbound_mod.c:261]: 
decode_flow_token(): flow-token failed validation
{1 322 BYE 218565972_26748892@x.x.x.x} INFO: rr [loose.c:519]: 
process_outbound(): failed to decode flow token
{1 322 BYE 218565972_26748892@x.x.x.x} INFO: rr [loose.c:794]: after_loose(): 
failed to process outbound flow-token

Is it possible to make the flow token survive a restart?

With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no
Mobile: 41 88 88 00


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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio

[Pan Christensen]

Dear Steve,

Would you mind sharing your findings and solution with the list?

With kind regards
Pan B. Christensen
Developer
Phonect AS 

> -Original Message-
> From: sr-users  On Behalf Of Wilkins,
> Steve
> Sent: mandag 11. juni 2018 12:30
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone
> when using Kamailio...works without Kamailio
> 
> Got it working.  Thank you everyone.
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
> Wilkins, Steve
> Sent: Sunday, June 10, 2018 3:06 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone
> when using Kamailio...works without Kamailio
> 
> >>>Email originates from a non-MITRE system. Use caution.<<<
> 
> Alex, Pan, Daniel,...
> Could this group => group:BUNDLE audio video in Message Body have
> anything to do with my Kamailio Video issue.
> 
> Thank you!!
> -Steve
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
> Wilkins, Steve
> Sent: Sunday, June 10, 2018 12:14 PM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone
> when using Kamailio...works without Kamailio
> 
> Hi Alex,
> No, I'm not using rtpengine.  It must definitely be some sort of Codec issue
> though, since I can seem to move the problem around and actually finally get
> video on the  softphone.  It was a little strange that I lost Audio on the
> WebRTC client though, since I only disabled VP8.
> 
> Since Kamailio is only relaying I am so confused why introducing Kamailio is
> messing up Video.
> 
> Thank you,
> -Steve
> 
> -Original Message-
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of
> Alex Balashov
> Sent: Sunday, June 10, 2018 11:46 AM
> To: Kamailio (SER) - Users Mailing List 
> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone
> when using Kamailio...works without Kamailio
> 
> Unless you're using rtpengine, you're just moving the problem around.
> Kamailio does nothing with SDP unless by way of rtpengine or told in some
> other way, such as sdpops.
> 
> On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" 
> wrote:
> >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I
> >now get Video on the softphone however, I lost two-way Audio.  Kamailio
> >seems to be  doing something with the codecs but still can't put my
> >finger on it.  The WebRTC Client, who is the caller, needs VP8 for
> >Video, and apparently Audio.
> >
> >I'm not sure if I getting closer or just moving the problem around.
> >
> >Thanks All!
> >
> >From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf
> >Of Wilkins, Steve
> >Sent: Saturday, June 9, 2018 12:47 PM
> >To: Kamailio (SER) - Users Mailing List 
> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when
> >using Kamailio...works without Kamailio
> >
> Email originates from a non-MITRE system. Use caution.<<<
> >I am desperately trying to resolve a big issue I have when using
> >Kamailio 5.2.0 and Asterisk 15.3.
> >
> >As soon as I go through Kamailio, I get no Video on either side of the
> >call; I do get two-way audio and the call stays connected.  If I simply
> >go right through Asterisk, I also get two-way Video.  I am having a
> >tough time determining why Kamailio is messing with the Video portion
> >of the call.
> >I have included a full pcap file.  As a side note when comparing pcap
> >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from
> >Asterisk and Softphone are exactly the same.
> >
> >Thank you,
> >-Steve
> 
> 
> -- Alex
> 
> --
> Sent via mobile, please forgive typos and brevity.
> 
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Pan Christensen]

I have to admit that I don't have much experience with video calls, but as far 
as I can tell VP8 is negotiated with WebRTC client. Nothing is negotiated with 
softphone.

WebRTC client offers:
Media Description, name and address (m): video 55185 UDP/TLS/RTP/SAVPF 96 97 98 
99 100 101 102 123 127 122 125 107 108 109 124

Asterisk offers to softphone:
Media Description, name and address (m): video 15112 RTP/AVP 99
Why is it advertising only 99?

Softphone answers with:
Media Description, name and address (m): video 36140 RTP/AVP 115 113 34 31
No match with offer, video is not negotiated for this call leg.

Asterisk answers WebRTC with:
Media Description, name and address (m): video 10328 UDP/TLS/RTP/SAVPF 96 100
96 is chosen for this call leg.


With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: lørdag 9. juni 2018 03:59
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello again Pan and Thank You for looking at this!!  Here is the pcap for the 
call that is not showing video on either side of the call.  As I mentioned I do 
get two way Audio, the call stays connected, but just no Video.  The Video 
ports look to be correct.  Video packets appear to go out (at least packets are 
going out over the Video ports). The screen on the Caller (A WebRTC Client) and 
the Called (A Provider Phone) are both blank.  The WebRTC client is Registered 
in Kamailio

Note: when I sanitized the file using tcprewrite an extra RTP packet appeared 
(Unknown RTP packet version 0), this was not here in the original.

I use Kamailio 5.2.0 and Asterisk 15.3

Thank you again!,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 5:12 PM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Dear Steve.

Asterisk is a B2BUA. It doesn't just route SIP messages like Kamailio does. 
Most of the time, it answers the call, sets up a new call to the other party 
and then bridges the two calls, making two separate call IDs (depending on your 
configuration).

Additional comments in blue inline below:


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  * Asterisk probably sends 183/200 back to callee, finalizing this codec 
negotiation. VP8 is chosen for this call leg.
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  * yes
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8
  * No. I assume that the callee didn't accept VP8 because it doesn't 
support it. It instead chose the highest priority codec it supported among the 
codecs that Asterisk advertised.
  * Asterisk accepts this codec because it supports it. There is no test 
checking which codec was chosen on the caller side and there is no 
renegotiation. Hence, video is broken forever.

Again, if you could show us the details (like attaching the SDPs that I asked 
for previously), then we wouldn't have to guess what's happening. We could tell 
you for sure.


Med vennlig hilsen
Pan B. Christensen

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 22:27
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi Pan,

If the Registered caller is a WebRTC client (Caller) whose preferred coded is 
VP8, calls a Phone (Called) whose preferred Video is H264, I should see


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8

Is this correct?  Then I think one of the issues is that there is no fmtp line 
in the VP8.  The only codecs that have fmtp lines is for the H264 codecs.

Thank you,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If y

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Pan Christensen]

Dear Steve.

Asterisk is a B2BUA. It doesn't just route SIP messages like Kamailio does. 
Most of the time, it answers the call, sets up a new call to the other party 
and then bridges the two calls, making two separate call IDs (depending on your 
configuration).

Additional comments in blue inline below:


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  * Asterisk probably sends 183/200 back to callee, finalizing this codec 
negotiation. VP8 is chosen for this call leg.
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  * yes
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8
  * No. I assume that the callee didn't accept VP8 because it doesn't 
support it. It instead chose the highest priority codec it supported among the 
codecs that Asterisk advertised.
  * Asterisk accepts this codec because it supports it. There is no test 
checking which codec was chosen on the caller side and there is no 
renegotiation. Hence, video is broken forever.

Again, if you could show us the details (like attaching the SDPs that I asked 
for previously), then we wouldn't have to guess what's happening. We could tell 
you for sure.


Med vennlig hilsen
Pan B. Christensen

From: sr-users  On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 22:27
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi Pan,

If the Registered caller is a WebRTC client (Caller) whose preferred coded is 
VP8, calls a Phone (Called) whose preferred Video is H264, I should see


  *   INVITE from Caller to Kamailio (this is where the Caller is Registered) 
offering VP8
  *   Kamailio will relay the INVITE to Asterisk
  *   Asterisk will INVITE Called
  *   Called sends '200 OK' to Asterisk requesting H264
  *   Asterisk Sends '200 OK' back to Kamailio to be relayed to Caller 
accepting VP8

Is this correct?  Then I think one of the issues is that there is no fmtp line 
in the VP8.  The only codecs that have fmtp lines is for the H264 codecs.

Thank you,
-Steve

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 11:06 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the 
order they are sent), we can tell you exactly what happens. Failing that, I'd 
say that the culprit is Asterisk, which probably negotiates two different 
codecs without the ability to transcode.

With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packe

Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Pan Christensen]

That's not strange. The soft phone probably doesn't advertise support for VP8, 
so it's not chosen.
Calling from WebRTC, I assume that VP8 is the preferred codec and Asterisk 
accepts it. Asterisk then finds out that the soft phone doesn't support VP8 and 
negotiates a different codec with that client.

If you show us the SDPs of INVITE and 183/200 on both sides of Asterisk (in the 
order they are sent), we can tell you exactly what happens. Failing that, I'd 
say that the culprit is Asterisk, which probably negotiates two different 
codecs without the ability to transcode.

With kind regards
Pan B. Christensen
Developer
Phonect AS
From: sr-users  On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 15:09
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Yes, for example a WebRTC Client (VP8) calls a Soft-Phone (H264).  What is 
strange is that if it is the other way around and the Soft-Phone calls the 
WebRTC client, it works.

Thank you

From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Pan 
Christensen
Sent: Friday, June 8, 2018 9:00 AM
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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Re: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

[Pan Christensen]

Hello Steve.

Does Asterisk negotiate different codecs with each client? If so, it needs to 
transcode, which I believe is currently not supported for video. What does 
Asterisk send back to device A?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: fredag 8. juni 2018 14:03
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Peculiar Kamailio Asterisk behavior on outbound calls

Hi All,

Issue: when a Call is made through Kamailio and Asterisk.  Asterisk uses 
incorrect Video RTP Payload Type when sending Video packets.

I have a situation where I make a call from Device A to Device B and Device A 
is Registered in Kamailio.  When Device A Calls Device B, Kamailio sends an 
'INVITE' to Asterisk, Asterisk then  'INVITES' Device B.  I get two-way Audio, 
the call stays connected, however, when Video packets are sent to Device B, the 
RTP Payload Type is incorrect.  The port is correct, but just not the Payload 
Type.

Here is where I think Kamailio is involved. In the first Invite from Kamailio 
to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, 
Device B wants to use '115 H264' and when Asterisk sends out Video packets, it 
is using '100' instead of '115', and of course I have no Video.  I don't know 
if this is just a coincidence but it sure seems like that is where the issue 
may lie.

Has anyone ever seen this behavior?  The Asterisk teams does not think it's an 
Asterisk issue.

Thank you,
-Steve
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[SR-Users] (N)DB_FIREBASE module?

[Pan Christensen]

Greetings all.





We are currently developing web and mobile apps using firebase: 
https://firebase.google.com/products/realtime-database/ . We have been 
discussing the possibility of Kamailio querying this database.



I assume that it will be possible to query it using HTTP and JSON modules, but 
it would probably be easier to use the DB API.



Any plans to develop such a module?




With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no
Mobile: 41 88 88 00


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Re: [SR-Users] configuring websockts in kamilio

[Pan Christensen]

Hello.

The js client cannot know this information, and hence cannot send it. What 
you’re seeing is normal for SIP over websockets. It should just work. What 
issues are you experiencing?

With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no
Mobile: 41 88 88 00


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From: sr-users  On Behalf Of vinod mn
Sent: fredag 1. juni 2018 20:21
To: sr-users@lists.kamailio.org
Subject: [SR-Users] configuring websockts in kamilio

hi configured websockets on kamilio 5.1.3
I am trying to register the client based on webrtc client ,(sip.js)
 the registration is fine but i am not getting the correct address  if i run 
"kamctl ul show"

In the address filed  I want the 
sip:@:>
ex: sip:12345@127.0.0.1:12345

output shows:
{
  "jsonrpc":  "2.0",
  "result": {
"Domains":  [{
"Domain": {
  "Domain": "location",
  "Size": 1024,
  "AoRs": [{
  "Info": {
"AoR":  "1x",
"HashID": 1x,
"Contacts": [{
"Contact":  {
  "Address": "sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws",
  "Expires":  397,
  "Q":  1,
  "Call-ID":  "ma4feu2i43bnu6ophe4k78",
  "CSeq": 86,
  "User-Agent": "SIP.js/0.5.0-devel BB",
  "Received": "sip:X.X.X.X:56087;transport=ws",
  "Path": "[not set]",
  "State":  "CS_NEW",
  "Flags":  0,
  "CFlags": 192,
  "Socket": "tcp:X.X.X.X:8080",
  "Methods":  783,
  "Ruid": "uloc-5b116457-1dc1-03",
  "Instance": 
"",
  "Reg-Id": 1,
  "Server-Id":  0,
  "Tcpconn-Id": 259,
  "Keepalive":  1,
  "Last-Keepalive": 1527876355,
  "Last-Modified":  1527876355
}
  }]
  }
}
  ],
  "Stats":  {
"Records":  1,
"Max-Slots":  1
  }
}
  }]
  },
  "id": 18958
}

In the address filed  I want the 
sip:@:>
ex: sip:12345@127.0.0.1:12345

Please give me any suggestions on this, which webrtc client can i use to 
register to kamailio and able to receive on the client and to make outbound 
calls from the client.

--
Thanks and regards
Vinod.M.N
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Re: [SR-Users] General Kamailio with Asterisk (or another PBX) Opinion

[Pan Christensen]

Hello,

This answer is somewhat based on previous questions that you have asked.

> Who forwards Registrations to Asterisk or PBX, and who lets Kamailio
> maintain Registrations?

As Alex said, it depends on what you are trying to accomplish. In addition, I 
would say that it depends on the call scenarios you expect to see.

Kamailio is more efficient at processing and routing SIP messages, so I let it 
do as much of the work as possible. I would only forward calls to Asterisk in 
order to do the things that Kamailio cannot, like answer the call in order to 
play an automated message, an IVR, record DTMF etc. Kamailio can implement 
calling queues, but I chose to put those in Asterisk as well as it has 
additional functionality.

For a normal call from user A to user B, you don’t need to involve asterisk at 
all.

With kind regards
Pan B. Christensen
Developer
Phonect AS

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Re: [SR-Users] INFO: stun_parse_header: incomplete header of STUN message

[Pan Christensen]

Dear Daniel.

Thanks for your quick reply and fix.

The proper solution for us is to make the backend stop sending keepalives as 
they are not needed in this case. I've asked our supplier about this as I 
couldn't find an option for it. I'm assuming that the backend incorrectly 
detects NAT and then automatically sends these packets.

I'm not a C programmer, but the fix looks good to me.

With kind regards
Pan B. Christensen
Developer
Phonect AS
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[SR-Users] INFO: stun_parse_header: incomplete header of STUN message

[Pan Christensen]

Hello!


After proxying a REGISTER from a WebRTC client to our SIP backend, the backend 
starts sending keepalives. Every 30 seconds I get a log entry in Kamailio 
saying: "INFO: stun [kam_stun.c:169]: stun_parse_header(): INFO: 
stun_parse_header: incomplete header of STUN message".

Here's one such packet:

No. Time   SourceDestination   Protocol 
Length Info
  1 0.00   xxx.xxx.xxx.xxx   yyy.yyy.yyy.yyyUDP  60 
5060 ? 15060 Len=4

Frame 1: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: 02:81:64:58:cf:d1 
(02:81:64:58:cf:d1)
Internet Protocol Version 4, Src: xxx.xxx.xxx.xxx, Dst: yyy.yyy.yyy.yyy
0100  = Version: 4
 0101 = Header Length: 20 bytes (5)
Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT)
Total Length: 32
Identification: 0x5bf5 (23541)
Flags: 0x4000, Don't fragment
Time to live: 56
Protocol: UDP (17)
Header checksum: 0x2fc3 [validation disabled]
[Header checksum status: Unverified]
Source: xxx.xxx.xxx.xxx
Destination: yyy.yyy.yyy.yyy
User Datagram Protocol, Src Port: 5060, Dst Port: 15060
Source Port: 5060
Destination Port: 15060
Length: 12
Checksum: 0xfa28 [unverified]
[Checksum Status: Unverified]
[Stream index: 0]
Data (4 bytes)

  00 00 00 00   
Data: 
[Length: 4]


I found this in kam_stun.c:

static int stun_parse_header(struct stun_msg* req, USHORT_T* error_code)
{

if (sizeof(req->hdr) > req->msg.buf.len) {
/* the received message does not contain whole 
header */
LOG(L_INFO, "INFO: stun_parse_header: 
incomplete header of STUN message\n");
/* Any better solution? IMHO it's not possible 
to send error response
* because the transaction ID is not available.
*/
return FATAL_ERROR;
}
...

Could someone please explain what's wrong?


Med vennlig hilsen
Pan B. Christensen
Utvikler

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-post: pan.christen...@phonect.no
Mobil: 41 88 88 00


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Re: [SR-Users] Question about using Kamailio and Asterisk and flow of an "INVITE"

[Pan Christensen]

Hello Steve.

What are you trying to achieve?

The call could go from client A to Kamailio to client B. No need to involve 
Asterisk. If you need PBX functionality, the INVITE needs to be routed to 
Asterisk, which will most likely answer the call and then set up a new call to 
client B. As Asterisk doesn't know where client B is, it needs to route this 
new call to Kamailio where client B is registered. It's possible for Asterisk 
to know where client B is but that solves nothing and may create other problems.

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Wilkins, Steve
Sent: onsdag 9. mai 2018 19:15
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] Question about using Kamailio and Asterisk and flow of an 
"INVITE"

Hello All,

I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using 
Kamailio and Asterisk.

Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the 
Invite back to Kamailio.  Also depending on
The version of Asterisk, the INVITE will then get forwarded to the AOR that is 
registered in Kamailio for the called number.
Does this seem correct?  It seems like there is an extra hop in there.

The reason I am now very curious now is because everything works fine if using 
Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3
I get the extra hop and call is dropped after 30 seconds.

I would appreciate any thoughts on this.

Thank you in advance.



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Re: [SR-Users] Understanding port changing

[Pan Christensen]

Hello!

This is probably caused by SIP ALG in the NAT router. Compare what the client 
sends to what you receive on the server.

With kind regards
Pan B. Christensen
Developer
Phonect AS

> -Original Message-
> From: sr-users  On Behalf Of Social
> Boh
> Sent: onsdag 9. mai 2018 18:32
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] Understanding port changing
> 
> Hello list,
> 
> I'm looking for to understand a problem have with Kamailio.
> 
> The configuration is:
> 
> user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT)
> 
> When I make a call from user A to user B (both registered) , User B receive
> the INVITE correctly but in the ringing SIP message change the port present
> on the Contact Header and same think on 200OK so when Kamailio try to
> send it ACK SIP message to this port fail because the Softphone listen on the
> port where has received the INVITE from Kamailio.
> 
> This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same
> computer.
> 
> Thank you
> 
> Regards
> 
> --
> ---
> I'm SoCIaL, MayBe
> 
> 
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Re: [SR-Users] WebRTC to SIP gateway

[Pan Christensen]

Hello again!

I've figured out that our SIP backend doesn't like headers with ';transport=ws' 
or Via: SIP/2.0/AUTO

I need to modify or remove contact, Via and Record-route headers.

I found that the TOPOS module is able to do this. I added it with default 
settings and applied it to all messages going to the backend. I was then able 
to make an outgoing call, but found that topology was only hidden in initial 
requests (not ACK, UPDATE, BYE etc.). Applying the module to messages in all 
directions worked better.
Being able to make outgoing calls is good progress, but this client will mostly 
be used for incoming calls (switchboard).
TOPOS does not modify REGISTER. I'm not sure if it's able to or if it should.
We tried modifying the headers in the client, but then Kamailio replies '478 
Unresolvable destination' to INVITE from the backend.
How can I enable the WebRTC client to register against the backend and maintain 
correct routing? Should TOPOS be able to do this? Can I use another module? Is 
it possible to register the client in Kamailio and then register Kamailio with 
the backend (building a new REGISTER message)? Any other suggestions?

With kind regards
Pan B. Christensen
Developer
Phonect AS

From: sr-users  On Behalf Of Pan 
Christensen
Sent: onsdag 9. mai 2018 15:09
To: sr-users@lists.kamailio.org
Subject: [SR-Users] WebRTC to SIP gateway

Hello!

It's been several years since I've used Kamailio. My current employer wants to 
implement WebRTC, which is currently not supported in our SIP backend, and 
asked if I could set up a Kamailio server as a gateway.

I've been able to make calls in all directions between SIP and WebRTC clients 
registered locally on Kamailio. When I tried to connect the server to the SIP 
backend, I ran into an issue. I'm able to register SIP clients in the backend 
via the gateway and make calls everywhere. However, the WebRTC client fails to 
register. Here are the messages between the Kamailio gateway and the SIP 
backend:


U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060
REGISTER sip:DOMAIN SIP/2.0.
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
Max-Forwards: 68.
To: .
From: ;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Contact: 
;+sip.ice;reg-id=1;+sip.instance="";expires=600.
Expires: 600.
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO.
Supported: path,gruu,outbound.
User-Agent: JsSIP 3.2.9.
Content-Length: 0.
Path: .
.


U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060
SIP/2.0 400 Wrong transport. Provided transport either invalid or not 
supported..
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
To: ;tag=91334f57.
From: ;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Content-Length: 0.


I believe that this error message is caused by ';transport=ws' in the Contact 
header. I'm not allowed to modify this header.

In the backend database, I found that some other clients have ';transport=UDP' 
in their path headers, so I tried to add that. (Why can I not add parameters in 
path module without adding username?) I still got the same error.

How do I best proceed?

For your information: We have outsourced the development of the WebRTC client, 
so we are able to change it. We also have the option of paying the supplier of 
the backend for development there.


With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no<mailto:pan.christen...@phonect.no>
Mobile: +47 41 88 88 00


 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/>
[facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] 
<https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Acompany%2CentityType%3AentityHistoryName%2CclickedEntityId%3Acompany_company_44983%2Cidx%3A0>

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[SR-Users] WebRTC to SIP gateway

[Pan Christensen]

Hello!

It's been several years since I've used Kamailio. My current employer wants to 
implement WebRTC, which is currently not supported in our SIP backend, and 
asked if I could set up a Kamailio server as a gateway.

I've been able to make calls in all directions between SIP and WebRTC clients 
registered locally on Kamailio. When I tried to connect the server to the SIP 
backend, I ran into an issue. I'm able to register SIP clients in the backend 
via the gateway and make calls everywhere. However, the WebRTC client fails to 
register. Here are the messages between the Kamailio gateway and the SIP 
backend:


U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060
REGISTER sip:DOMAIN SIP/2.0.
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
Max-Forwards: 68.
To: .
From: ;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Contact: 
;+sip.ice;reg-id=1;+sip.instance="";expires=600.
Expires: 600.
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO.
Supported: path,gruu,outbound.
User-Agent: JsSIP 3.2.9.
Content-Length: 0.
Path: .
.


U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060
SIP/2.0 400 Wrong transport. Provided transport either invalid or not 
supported..
Via: SIP/2.0/UDP 
GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0.
Via: SIP/2.0/AUTO 
lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151.
To: ;tag=91334f57.
From: ;tag=9qhqhrnj3s.
Call-ID: j5h7830ivr5dfc2mn5sov1.
CSeq: 4 REGISTER.
Content-Length: 0.


I believe that this error message is caused by ';transport=ws' in the Contact 
header. I'm not allowed to modify this header.

In the backend database, I found that some other clients have ';transport=UDP' 
in their path headers, so I tried to add that. (Why can I not add parameters in 
path module without adding username?) I still got the same error.

How do I best proceed?

For your information: We have outsourced the development of the WebRTC client, 
so we are able to change it. We also have the option of paying the supplier of 
the backend for development there.


With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no
Mobile: +47 41 88 88 00


 [cid:image007.png@01D3A0E8.376921D0] 
[facebook_2] [LinkedIn_logo_initials (1)] 


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