Colin,
There is gratitude for you for your time and exceptional consideration.
I am really indebted, mate! Here in Albanian culture "I owe you a coffee"
(no a dinner!) :)
So as per your advice, I shall stick to the run of the mill clients at this
stage, rather than reinvent a slightly better whee
Zakia,
Alright, starting with the basics: IP based voice/video communications
generally involve two distinct responsibilities: signaling and media.
The job of the signaling layer is to exchange information about calls
between two parties. For example, if I call you, I use my signaling
protocol to
Hello Colin:
Thanks a lot! It solved the issue. Though I had got a hint by succeeding to
specify the address using *kamailio -l desired address/port *option, yet
never noticed the lack of *MY-WS_ADDRESS*
Last, but not the least, in a browser supporting WEBRTC (FF, CHROME) do we
still need SIP ML5
Zaka,
I could be wrong here but I don't think you ever actually have a "listen"
line for MY_WS_ADDR.
I believe you have a typo, as you have listen=MY_IP_ADDR twice, once within
the guard for WITH_WEBSOCKETS. Replace the one inside the if with
MY_WS_ADDR and I think your problem should be resolved
Dear List:
I am still stuck. Still unable to get Kamailio to listen on WEBRTC port*
(in this case 8000)*
Please advise if it has something to do with disabling TLS/MSRP?
Please see the log & related config file in line here:
*Log:*
0(5724) INFO: [ppcfg.c:82]: pp_subst_add(): ### added subst
Hello,
On 26/05/16 18:36, Arsen wrote:
> Hi Daniel,
>
> nope debug=3 doens't give more info.
does this mean that you don't see other log messages from tls or that
those messages don't give any useful detail?
> I have the same certificate on the web server and on the kamailio
> (same crt/key on b
Hi Daniel,
nope debug=3 doens't give more info.
I have the same certificate on the web server and on the kamailio (same
crt/key on both)
Thanks in advance
On Thu, May 26, 2016 at 5:58 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> if you run with debug=3, do you get more hints from the debug
Hello,
if you run with debug=3, do you get more hints from the debug messages?
I guess you require client certificate in your config.
Cheers,
Daniel
On 26/05/16 15:06, Arsen wrote:
> Hi guys!
>
> I am trying to configure kamailio with WSS.
> We have trusted certificate installed SIP over TCP/T
Hi guys!
I am trying to configure kamailio with WSS.
We have trusted certificate installed SIP over TCP/TLS works fine.
But when I try WSS I got error:
ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL
routines:SSL3_READ_BYTES:tlsv1 alert access denied
ERROR: [tcp_read.c:
Great Daniel! Problem solved.
Thanks,
Moacir
To: sr-users@lists.sip-router.org
From: mico...@gmail.com
Date: Wed, 18 May 2016 07:08:02 +0200
Subject: Re: [SR-Users] WebRTC
Hello,
if you don't have a trusted certificate, then browse first to
https://kamailioip
Hello,
if you don't have a trusted certificate, then browse first to
https://kamailioip:5061 (or your wss port) and accept the certificate.
If not working, maybe we can figure out what is the issue if you post
the logs with debug=3 here.
Cheers,
Daniel
On 16/05/16 18:11, Moacir Ferreira wrote:
You can start with websockets module documentation. It gives details
explaination of how it works and what needs to be done in config.
http://kamailio.org/docs/modules/4.4.x/modules/websocket.html#idp1214176
I have worked with JSSIP, SIPML5 and a few others using Kamailio and
FreeSWITCH, all work
Hi,
I am trying to use Kamailio with WebTRC to make and receive calls from the
browser.
Using the rpms from the Kamailio repository I have installed and tried the
websocket config example from the source code. Using the first debug example
from here https://www.kamailio.org/wiki/tutorials/tls
Hi,
I am trying to use Kamailio with WebTRC to make and receive calls from the
browser.
Using the rpms from the Kamailio repository I have installed and tried the
websocket config example from the source code. Using the first debug example
from here https://www.kamailio.org/wiki/tutorials/tls
I already use DTLS-SRTP (websockets dont works with RTP).
This is my SDP body. And I have no sound at incoming calls
tcpdump shows me that I have no rtp strean fro websocket endpoint
v=0
o=root 1828066564 1828066564 IN IP4 1.1.1.1
s=Cattaxi Media Server
c=IN IP4 1.1.1.1
t=0 0
m=audio 30328 RTP/SA
Hi!
use DTLS-SRTP, to say how to handle it with rtpengine - I think you should
provide more info about your setup, and call cases
Cheers!
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/WebRTC-no-longer-supports-RTP-tp143834p143836.html
Sent from the Users mailing list
https://developers.google.com/web/updates/2015/10/chrome-47-webrtc
So at 47 chrome we already have no sound.
What kind of proto we must use and how to handle this with rtpengine?
Do anyone have same problems with it?
___
SIP Express Router (SER) and Kama
Hello Giovanni,
I fixed webrtc call working right now. I checked debug again and I so 408
error. There was missing record_route() for first INVITE.
Thank you for help.
Slava.
From: "Giovanni Maruzzelli"
To: "sr-users"
Sent: Saturday, 5 September, 2015 07:50:29
Su
Hello Giovanni,
Here SDP webrtc call traffic capture on kamailio side
http://fpaste.org/264289/15798761/
password from paste: kamailio
Slava.
From: "Giovanni Maruzzelli"
To: "sr-users"
Sent: Saturday, 5 September, 2015 07:50:29
Subject: Re: [SR-Users] webrtc
First thing, check that you can have a webRTC call directly with
FreeSWITCH, without passing by Kamailio.
If that works, and it does not work through Kamailio, then provide kamailio
config, freeswitch config, and debug log from both.
On Sat, Sep 5, 2015 at 12:51 AM, Slava Bendersky
wrote:
> H
Hello Everyone,
I am trying setup webrtc call with freeswitch.
[root@canlvprx01 kamailio]# rpm -qa | grep kamailio
kamailio-4.3.1-4.4.fc21.x86_64
Case:
INTERNET eth0 | eth1 LAN
|
Web Browser >webrtc > kamailio > freeswitch
Getting this error
Sep 4 00:22:06 canlvprx01 /us
Hi,
I have configured kamailio 4.3.1 with mysql ,rtpproxy,websocket modules ,
then I tried to call through the two extensions using x-lite phone and
get the capture. Call is connected.
In that test rtp was going through the rtpproxy server,
Then I configured jssip to test webrtc in this setu
Hello,
kamailio + rtpengine can be used for webrtc calls between browsers as
well as browser to classic sip phones. You can fine on github some
config examples, published by Carlos Ruiz Diaz.
Using this combination you can place an instance in front of asterisk
and let asterisk behave as a classi
I made successful audio calls from browser to browser using Asterisk
13.1 and SIPML5 browser phone.
Asterisk can't manage WebRTC video calls due to lack of codec
negotiation module, but I also faced RTP ports NAT traversal issue. To
my understanding Kamailio is capable to resolve this.
Can anybo
Thank you Marc.
On Thu, Feb 12, 2015 at 11:51 PM, Marc Soda wrote:
> Our config is based on the example config and the WebRTC bits are based on
> Carlos'.
> I've attached the relevant parts. It's pretty heavily customized to our
> specific environment. The main differences are the way that w
Our config is based on the example config and the WebRTC bits are
based on Carlos'.
I've attached the relevant parts. It's pretty heavily customized to our
specific environment. The main differences are the way that we detect a
video call, how we route to our backend servers, and that we send vi
Gentle Reminder !
Thanks
Warm Regds,
Rahul
On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR
wrote:
> Thanks guys !
>
> I did further investigation of the Chrome logs and found that... (this is
> really interesting), even though I disabled Video; still JSsip was sending
> video information in the
Thanks guys !
I did further investigation of the Chrome logs and found that... (this is
really interesting), even though I disabled Video; still JSsip was sending
video information in the m & a lines.
The fact that I was trying to call PSTN number made it mandatory to set
video port to '0' in 183
We are in the middle of designing a similar solution with Kamailio and
rtpengine and after some initial problems things are going really well. I
can tell you that we ended up going with SIPjs over JSSip and it handled a
lot of the weird browser specific issues we were having.
I'm not sure about t
Maybe you could include you config also?
On 10 February 2015 at 15:01, Rahul MathuR wrote:
> Hello gents,
>
> I was trying my hands on getting a successful RTCweb call (JSsip, since
> Peter Dunkley mentioned that he's been using JSsip for most of the testing
> scenarios..) to PSTN, making my kam
Richard Fuchs writes:
> In the third case, the audio stream was setup as a=sendonly, explaining
> the one-way audio. Probably caused by Firefox not being able to access
> the playback device.
as i reported in another message, i have also lately noticed in my tests
that firefox sends invite with a
On 12/18/14 13:38, Andrey Utkin wrote:
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
>
Also I encounter such issue: even in the working scenario, the hangup
of one peer doesn't make the call end for another peer.
rtpengine: https://gist.github.com/krieger-od/1cfe84b53dc0d29cfb90
kamailio: https://gist.github.com/krieger-od/11c6bbf7dad15382e81b
ngrep: https://gist.github.com/krieger-o
2014-12-18 20:38 GMT+02:00 Andrey Utkin :
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
This doesn't work: few seconds after answer, there's
2014-12-18 20:09 GMT+02:00 Andrey Utkin :
> 2014-12-18 20:05 GMT+02:00 Richard Fuchs :
>> Amazon NAT is exactly why I've mentioned it, because on an Amazon
>> system, if you don't use the --interface option correctly
>> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
>> that
2014-12-18 20:05 GMT+02:00 Richard Fuchs :
> Amazon NAT is exactly why I've mentioned it, because on an Amazon
> system, if you don't use the --interface option correctly
> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
> that show in your log.
Thank you, will try with suc
On 12/18/14 12:55, Andrey Utkin wrote:
> 2014-12-18 19:30 GMT+02:00 Richard Fuchs :
>> Write error on RTP socket usually indicates an incorrect network setup,
>> for example trying to use a source address for IP packets which isn't
>> bound to any local network interface (especially if you're sitti
2014-12-18 19:30 GMT+02:00 Richard Fuchs :
> Write error on RTP socket usually indicates an incorrect network setup,
> for example trying to use a source address for IP packets which isn't
> bound to any local network interface (especially if you're sitting
> behind NAT), or local iptables rules re
On 12/18/14 12:11, Andrey Utkin wrote:
> Hi!
> I need to establish calls between WebRTC and usual SIP clients
> (exactly, sipml/jssip and linphone-android).
> I used configs from https://github.com/caruizdiaz/kamailio-ws and
> latest git master HEAD of both kamailio and
> rtpengine. I got calls fro
Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only wit
The main thing you need to look out for is that your registrar supports the
Path and Outbound specifications in order to correctly route INVITEs to
your WebSocket clients via the edge proxy. I'm in a situation right now
where I'm having some difficulty getting a Kamailio WebSocket edge proxy
playin
Hi,
> a) Can kamailio be used as sip-proxy while using WebRTC based UA
> calling to plain UAC/WebRTC based UAC ?
Yes, kamailio can do SIP over websocket, so all you need is a
javascript SIP stack (e.g. JsSIP, jain-sip JS, ...) on your WebRTC
enabled client.
> b) What to use for media proxying (t
Hello,
I'm new to WebRTC although I've been using kamailio as sip proxy server for
few months now. What I really do not know and trying to understand is -
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to
plain UAC/WebRTC based UAC ?
b) What to use for media proxying (t
I'll be in Norfolk, VA for xTupleCon in October
On 15 October, there will be two events for WebRTC:
14:15 a talk about the xTuple WebRTC extension at xTupleCon
- must register for xTupleCon to attend this
17:30 a technical / developer workshop at xTuple's offices
- free, anybody welcome,
Use rtpengine for this. You may use rtpproxy-ng module to manipulate
options of rtpengine.
20.08.2014 11:07 пользователь написал:
> Hi,
>
> We are using Kamailio as a WebRTC proxy. We have converted the signaling
> successfully.
>
> Now, for media, is it possible to convert srtp to rtp using rtpp
Hi,
We are using Kamailio as a WebRTC proxy. We have converted the signaling
successfully.
Now, for media, is it possible to convert srtp to rtp using rtpproxy_ng or
mediaproxy? If yes can you provide me with some details?
Thanks in advanced!
Sent from my “contract free” BlackBerry® smartphon
Hi all, Another attempt,
After doing some tests, I saw that one of the problems was that was
necessary to comment the following lines within the deffinition of the
RELAY route:
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
#if (i
Apologize. Previous message was too long.
L.
El 02/06/2014 20:25, "LAA" escribió:
> Hi all,
>
> Another guy strugling his mind trying to get a configuration to enable
> calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone)
> I've been working with the examples that were shared
You might need to also add asterisk 12 b2b in order to convert to simple
sip to solve issues with ice on the same box.
On Apr 1, 2014 11:52 AM, "ik" wrote:
> Hello,
>
> I'm a newbie with Kamailio, and I require to connect webrtc (websockets)
> based phones, into a VoIP PBX that does not support w
Hello,
I'm a newbie with Kamailio, and I require to connect webrtc (websockets)
based phones, into a VoIP PBX that does not support websockets.
I wish to create/use Kamailio rules that will translate UDP to websockets
and vice versa.
I have found few examples over the internet, but as it seems (
just an update to this issue, this was caused by rtpproxy not answering to
a stun binding request but rather relaying it downstream. (for which
asterisk as pstn terminator replied with a binding success)
i wonder how difficult it is to add stun into rtpproxy? mediaproxy-ng
should work but it does
here is the scenario:
kamailio 4.0.4 running in bridge mode
eth0 : 10.17.0.202
eth1 : 100.200.30.40
a call comes in from ice enabled webrtc client (chrome) from public
internet,
kamailio processed the call, no audio.
if i replace kamailio with asterisk, audio ok.
upon investigating using wireshar
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