Hello Sirs, Sir Richard,
This is working perfect. I have tried the following test cases:
1. WS - WS in intranet; PASSED
2. WS - WS in different networks, both with restrictive firewalls: PASSED
3. WS - SIP Client (Boghe, Jitsi) in intranet: PASSED
4. WS - IMSDroid, both with restrictive firewalls:
On 02/22/14 07:07, Mihai Marin wrote:
> Hello Sirs, Sir Richard,
> Thank you for your detailed explication.
> I'm still thinking on that but I would say to act as the caller and keep
> caller decision. If caller makes an offer with rtcp-mux ,
> include separate ICE candidates for RTCP for media pro
Just in case someone is interested, I created a sample script that could
help new comers having the same problem.
I will write a blog entry explaining how this works, but in a nutshell:
- this script is configured to run behind NAT, port TCP 10080 and TCP/UDP
5090 are exposed to the Internet
- yo
On 02/22/14 07:07, Mihai Marin wrote:
> Hello Sirs, Sir Richard,
> Thank you for your detailed explication.
> I'm still thinking on that but I would say to act as the caller and keep
> caller decision. If caller makes an offer with rtcp-mux ,
> include separate ICE candidates for RTCP for media pro
Hello Sirs, Sir Richard,
Thank you for your detailed explication.
I'm still thinking on that but I would say to act as the caller and keep
caller decision. If caller makes an offer with rtcp-mux , include separate
ICE candidates for RTCP for media proxy too and forward as it is to alice.
If callee
On 02/20/14 04:15, Mihai Marin wrote:
> Hello Sirs, Sir Richard,
> I understand the problem but I don't understand the behavior. Let me
> tell you how I understood the problem and where I misunderstand the
> behavior.
>
> BOB sens an offer to Alice with rtcp-mux. The flow is: UAC (bob) -
> Kamaili
Hello Sirs, Sir Richard,
I understand the problem but I don't understand the behavior. Let me tell
you how I understood the problem and where I misunderstand the behavior.
BOB sens an offer to Alice with rtcp-mux. The flow is: UAC (bob) - Kamailio
- MP-NG - Kamailio - UAS (alice). From the offer,
On 02/18/14 11:12, Mihai Marin wrote:
> Hello Sirs,
> Thank you, one step forward but still buggy - half buggy :)
>
> Now, it's working just one way. If bob calls alice, alice will receive
> video but bob won't. If I stop mediaproxy-ng process (without any other
> modification) and redo the call,
Hello Sirs,
Thank you, one step forward but still buggy - half buggy :)
Now, it's working just one way. If bob calls alice, alice will receive
video but bob won't. If I stop mediaproxy-ng process (without any other
modification) and redo the call, everything is just fine.
The invite sdp that alic
On 02/12/14 04:41, Mihai Marin wrote:
> Hello,
> I managed to try it out and I have good news and bad news :)
>
> The good news is that always TURN is working perfect. So, if I remove
> all the ice-candidates (rtpproxy_manage("+")) everything is perfect.
That's good to hear!
> If
> I just append
Hello,
Would you mind posting your kamailio configuration file please?
Thanks,
Ashu
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Hello,
I managed to try it out and I have good news and bad news :)
The good news is that always TURN is working perfect. So, if I remove all
the ice-candidates (rtpproxy_manage("+")) everything is perfect. If I just
append turn candidate (rtpproxy_manage()) I have strange errors in kamailio
log a
Hello,
Hope you don't mind if I borrow this topic to place a question or request.
In past days I successfully setup Kamailio in my local network and made
successful WebRTC to WebRTC and SIP to SIP calls. The problem is with WebRTC
to SIP call. I also added mediaproxy-ng by following instructions f
On 02/06/14 14:42, Muhammad Shahzad wrote:
> Great, i would test Bundle right away. Just wondering if this branch
> also supports DTLS--SRTP. I would love to test that feature when available.
Not quite yet, but it's being implemented as we speak.
cheers
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Great, i would test Bundle right away. Just wondering if this branch also
supports DTLS--SRTP. I would love to test that feature when available.
Thank you.
On Thu, Feb 6, 2014 at 2:43 PM, Richard Fuchs wrote:
> Hey,
>
> What mediaproxy-ng can do (which other RTP proxies don't do) is
> transl
Hey,
What mediaproxy-ng can do (which other RTP proxies don't do) is
translate RTP/AVP (from regular SIP endpoints) to and from RTP/SAVPF
(encrypted RTP from WebRTC), which is what people usually use it for.
Assuming that ICE does its job correctly, two WebRTC endpoints should be
able to communic
Hello,
I'm trying the simplest case first. I don't understand why you are saying
that most of the people don't use mediaproxy-ng for WebRTC to WebRTC calls.
If my firewall is a restrictive one I need to use turn server and
mediaproxy-ng can do turn too? Probably I'm not seeing the big picture.
Reg
Hey,
If you're trying to connect two WebRTC endpoints with each, you don't
need any of mediaproxy-ng's magic to get it working. All the previous
replies were assuming that you were trying to connect a WebRTC endpoint
with a non-WebRTC one, which is usually what people are trying to do.
In your ca
Hello,
Thank you for your detailed explication but I miss some information or I'm
unable to understand it properly. What I'm trying to do is to use
mediaproxy-ng as a turn server between 2 WebRTC endpoints (when at least
one is behind restrictive firewall). Trying to replicate what you explained
on
There are several problems that need to be addressed in your kamailio.cfg
but let me try to focus only on mediaprxoy-ng related ones.
First instead of engaging mediaproxy in failure route, engage it main route
or branch route. Why wait for failure when we know call will fail anyway if
you try to c
Hi,
could you please post also your Chrome js developer log?
Does the problem exist if you start the jssip clients without video support?
Andrew
On 02/03/2014 12:00 PM, Mihai Marin wrote:
> Hello,
>
> Another weekend struggling to make a call from jssip to another jssip
> behind firewall and I s
Peter had a talk at Astricon 2013 presenting how it works. I think the
magic lies on the parameters. See this slide (there are some more
interesting slides)
https://www.youtube.com/watch?list=PLighc-2vlRgQHZMBp-8CCFi5otCnw7Lwj&v=rXsVSaRuv20&feature=player_detailpage#t=659
regards
Klaus
On 29
Hi!
The problem is different SDP formats between normal SIP
clients/gateways, and WebRTC clients.
You have to instruct mediaproxy-ng to rewrite the SDP and do the
conversion (encryption, ...).
So either the rtpproxy_ng calls lack the commands for the "gatewaying",
or the webrtc clients use
Hello Sirs,
I have a problem configuring kamailio with mediaproxy-ng and I'm asking for
help.
I managed to build everything, kamailio find support for mediaproxy-ng
using rtpproxy-ng. When I'm trying to make a call from Web using my phone's
internet provider to my computer's web I get 488 Not Acce
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