16 May
2016
16 May
'16
12:11 p.m.
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/testing-and-debugginga I can see that my TLS
wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to
register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing
some research, it is more like that several products are currently broken and not
working...
So can anyone help me (share a config and tricky) or point to a document that would help
me to get it running?
Thanks,
Moacir
16 May
16 May
12:42 p.m.
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 worked fine for me. So i guess the only problem is your
understanding and configuration of the setup NOT these products.
Thank you.
On Mon, May 16, 2016 at 6:11 PM, Moacir Ferreira <moacirferreira(a)hotmail.com
wrote:
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/testing-and-debugginga I can
see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I
get no success, they fail to register in Kamailio. From the logs I can see
that something goes wrong with TLS. Doing some research, it is more like
that several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a document
that would help me to get it running?
Thanks,
Moacir
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
18 May
18 May
1:08 a.m.
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:
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/testing-and-debugginga I
can see that my TLS wss setup is working. But I try JSCommunicator and
sipML5 I get no success, they fail to register in Kamailio. From the
logs I can see that something goes wrong with TLS. Doing some
research, it is more like that several products are currently broken
and not working...
So can anyone help me (share a config and tricky) or point to a
document that would help me to get it running?
Thanks,
Moacir
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
7:09 a.m.
Great Daniel! Problem solved.
Thanks,
Moacir
To: sr-users(a)lists.sip-router.org
From: miconda(a)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: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:
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/testing-and-debugginga
I can see that my TLS wss setup is working. But I try
JSCommunicator and sipML5 I get no success, they fail to
register in Kamailio. From the logs I can see that something
goes wrong with TLS. Doing some research, it is more like that
several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a
document that would help me to get it running?
Thanks,
Moacir
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
1:16 p.m.
New subject: Browser WebRTC transcoder
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to
legacy VoIP?
Moacir
3:18 p.m.
New subject: Browser WebRTC transcoder
Hello,
kamailio+rtpengine should do this job quite well.
Cheers,
Daniel
On 18/05/16 19:16, Moacir Ferreira wrote:
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow
browser WebRTC calls to legacy VoIP?
Moacir
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
4:57 p.m.
New subject: Browser WebRTC transcoder
Hey Daniel,
If you say so, you probably right... I did not try it because on the sipwise GitHub
(https://github.com/sipwise/rtpengine) they mention:
"Rtpengine does not (yet) support:
Repacketization or transcodingPlayback of pre-recorded streams/announcementsRecording of
media streamsZRTP"
So I did not test it... I will give it a try. By the way, from where should I download the
source code? Also, any "tricky" (common mistake like my last one on WebRTC TLS)
I should care about before trying it?
Thanks,
Moacir
To: sr-users(a)lists.sip-router.org
From: miconda(a)gmail.com
Date: Wed, 18 May 2016 21:18:01 +0200
Subject: Re: [SR-Users] Browser WebRTC transcoder
Hello,
kamailio+rtpengine should do this job quite well.
Cheers,
Daniel
On 18/05/16 19:16, Moacir Ferreira
wrote:
A question for the community:
What would be your best advice for a RTP proxy/transcoder to
allow browser WebRTC calls to legacy VoIP?
Moacir
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:03 p.m.
New subject: Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the
sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/
//
* /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to
PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
is supported.
Cheers
7:51 p.m.
New subject: Browser WebRTC transcoder
I did not dig into the problem but on my tests I saw that my (old) Grandstream phone was
refusing the call for not having a compatible codec to talk with the offered ones by the
browser (Firefox). Being this the case, I guess I must include a translator, and all
routing logic, in between the callers. It points to Asterisk that I would like to avoid
for now. But I guess this is not a problem that only affects me. Someone else must have
faced this before. So the question still open: What solution would be recommended for such
case?
Cheers,
Moacir
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Wed, 18 May 2016 19:03:10 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the
sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/
//
* /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to
PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
is supported.
Cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
19 May
19 May
12:22 a.m.
New subject: Browser WebRTC transcoder
What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers,
Daniel
On 19/05/16 01:51, Moacir Ferreira wrote:
I did not dig into the problem but on my tests I saw
that my (old)
Grandstream phone was refusing the call for not having a compatible
codec to talk with the offered ones by the browser (Firefox). Being
this the case, I guess I must include a translator, and all routing
logic, in between the callers. It points to Asterisk that I would like
to avoid for now. But I guess this is not a problem that only affects
me. Someone else must have faced this before. So the question still
open: What solution would be recommended for such case?
Cheers,
Moacir
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Wed, 18 May 2016 19:03:10 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Hey Daniel,
If you say so, you probably right... I did not try it because on the
sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/
//
* /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to
PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
is supported.
Cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:52 a.m.
New subject: Browser WebRTC transcoder
Yes G711 is offered... I guess the Grandstream phone is "confused" about the way
the SIP browser stack talks. I got the following logs taken on Kamailio, using ngrep:
######################### START OF LOG ##########################
# Call from Grandstream to browser
##################################
[root@sip ~]# ngrep -d any -qt -W byline port 5060
interface: any
filter: ( port 5060 ) and (ip or ip6)
...
U 2016/05/19 09:15:24.701195 192.168.100.85:5060 -> 192.168.100.159:5060
INVITE sip:1001@192.168.100.159 SIP/2.0.
Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.
From: "Fernando" <sip:1002@192.168.100.159>;tag=b0d53bed080e1b0f.
To: <sip:1001@192.168.100.159>.
Contact: <sip:1002@192.168.100.85:5060;transport=udp>.
Supported: replaces, timer, path.
P-Early-Media: Supported.
Proxy-Authorization: Digest username="1002", realm="192.168.100.159",
algorithm=MD5, uri="sip:1001@192.168.100.159",
nonce="Vz13SFc9dhwixIQaHqfrXSDvtNLkf+guJwNHi4A=",
response="9c4f2fc2b7f179172fe9a6adf0d2f60f".
Call-ID: dda078a035a57ecb(a)192.168.100.85.
CSeq: 8700 INVITE.
User-Agent: Grandstream BT200 1.2.5.3.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK.
Content-Type: application/sdp.
Content-Length: 385.
.
v=0.
o=1002 8000 8001 IN IP4 192.168.100.85.
s=SIP Call.
c=IN IP4 192.168.100.85.
t=0 0.
m=audio 11022 RTP/AVP 18 8 0 3 9 2 97 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=20.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
U 2016/05/19 09:15:24.710083 192.168.100.159:5060 -> 192.168.100.85:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.
From: "Fernando" <sip:1002@192.168.100.159>;tag=b0d53bed080e1b0f.
To: <sip:1001@192.168.100.159>;tag=56f8047e80cfcc9e90a3acc9609da1ba-9e91.
Call-ID: dda078a035a57ecb(a)192.168.100.85.
CSeq: 8700 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).
Content-Length: 0.
######################### END OF LOG ##########################
So the Grandstream offers a lot of codecs but will get a "Not Found" from
Kamailio. Look in the other way:
######################### START OF LOG ##########################
# Call from browser to Grandstream
##################################
[root@sip ~]# ngrep -d any -qt -W byline port 5060
interface: any
filter: ( port 5060 ) and (ip or ip6)
U 2016/05/19 09:25:11.285826 192.168.100.159:5060 -> 192.168.100.85:5060
INVITE sip:1002@192.168.100.85:5060;transport=udp SIP/2.0.
Record-Route: <sip:192.168.100.159;r2=on;lr=on>.
Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.
Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.
From: "Moacir"<sip:1001@my.lab>;tag=MhRTswgaXENAxcDi25HJ.
To: <sip:1002@my.lab>.
Contact:
"Moacir"<sips:1001@df7jal23ls0d.invalid;alias=192.168.100.249~59318~6;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr".
Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.
CSeq: 15365 INVITE.
Content-Type: application/sdp.
Content-Length: 1182.
Max-Forwards: 69.
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04.
Organization: Doubango Telecom.
.
v=0.
o=mozilla...THIS_IS_SDPARTA-46.0.1 947803314240298800 0 IN IP4 127.0.0.1.
s=Doubango Telecom - firefox.
t=0 0.
a=sendrecv.
a=fingerprint:sha-256
44:F1:6D:31:F4:D6:D9:43:1D:38:0B:8E:67:1E:5F:DD:10:F4:5F:1C:4B:7E:7A:47:F8:85:C4:93:40:A7:2D:5E.
a=ice-options:trickle.
a=msid-semantic:WMS *.
m=audio 61455 UDP/TLS/RTP/SAVPF 109 9 0 8.
c=IN IP4 192.168.100.249.
a=candidate:0 1 UDP 2122252543 192.168.100.249 61455 typ host.
a=candidate:1 1 UDP 2122187007 2001:0:5ef5:79fd:24be:2fcf:fa06:dbc8 61456 typ host.
a=candidate:0 2 UDP 2122252542 192.168.100.249 61457 typ host.
a=candidate:1 2 UDP 2122187006 2001:0:5ef5:7
U 2016/05/19 09:25:11.348673 192.168.100.85:5060 -> 192.168.100.159:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.
Via: SIP/2.0/WSS
df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.
Record-Route: <sip:192.168.100.159;r2=on;lr=on>.
Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.
From: "Moacir"<sip:1001@my.lab>;tag=MhRTswgaXENAxcDi25HJ.
To: <sip:1002@my.lab>;tag=9ea71e1d1f839cef.
Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.
CSeq: 15365 INVITE.
User-Agent: Grandstream BT200 1.2.5.3.
Warning: 304 GS "Media type not available".
Content-Length: 0.
######################### END OF LOG ##########################
Here the Grandstream says "Media type not available". As I am not a real SIP
guy, I got no clue why does not work!
Anyway, I am using the latest RPMs from Kamailo, running it using the websocket.cfg
suggested configuration, no rtpengine installed on it. At the WebRTC, I am using sipml5
configuring it not to use STUN/TURN.
Cheers!
Moacir
To: sr-users(a)lists.sip-router.org
From: miconda(a)gmail.com
Date: Thu, 19 May 2016 06:22:57 +0200
Subject: Re: [SR-Users] Browser WebRTC transcoder
What codecs are supported by your grandstream? Isn't the g711 in
the group?
Cheers,
Daniel
On 19/05/16 01:51, Moacir Ferreira
wrote:
I did not dig into the problem but on my tests I
saw that my (old) Grandstream phone was refusing the call for
not having a compatible codec to talk with the offered ones by
the browser (Firefox). Being this the case, I guess I must
include a translator, and all routing logic, in between the
callers. It points to Asterisk that I would like to avoid for
now. But I guess this is not a problem that only affects me.
Someone else must have faced this before. So the question still
open: What solution would be recommended for such case?
Cheers,
Moacir
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Wed, 18 May 2016 19:03:10 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
> Hey Daniel,
>
> If you say so, you probably right... I did not
try
it because on the
> sipwise GitHub
(https://github.com/sipwise/rtpengine) they mention:
>
> /"Rtpengine does not (yet) support:/
> //
>
> * /Repacketization or transcoding/
This refers to translating one audio codec into
another
(e.g. opus to
PCM). Translating between RTP and SRTP (i.e.
encrypting
and decrypting)
is supported.
Cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list
sr-users(a)lists.sip-router.org
8:22 a.m.
New subject: Browser WebRTC transcoder
On 05/19/2016 04:52 AM, Moacir Ferreira wrote:
...
So the Grandstream offers a lot of codecs but will get
a "Not Found"
from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a
missing registration.
Here the Grandstream says "Media type not
available". As I am not a real
SIP guy, I got no clue why does not work!
This you can solve with rtpengine. The required codecs (PCM) are there,
you just need to break the encryption (RTP <> SRTP) and some other
features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine
can do.
Cheers
20 May
20 May
3:49 a.m.
New subject: Browser WebRTC transcoder
Yes, now it is working both ways. I was missing the rtpengine and right configuration.
I will do some further testing to see if everything is functional.
Thanks,
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Thu, 19 May 2016 08:22:26 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 05/19/2016 04:52 AM, Moacir Ferreira wrote:
...
So the Grandstream offers a lot of codecs but
will get a "Not Found"
from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a
missing registration.
Here the Grandstream says "Media type not
available". As I am not a real
SIP guy, I got no clue why does not work!
This you can solve with rtpengine. The required codecs (PCM) are there,
you just need to break the encryption (RTP <> SRTP) and some other
features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine
can do.
Cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
2926
days inactive
2930
days old
12 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
M S -
Moacir Ferreira -
Richard Fuchs