27 Jan
2015
27 Jan
'15
3:39 a.m.
Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with
rtpengine from sipwise to handle media as evident from he logs.
This architectue uses a TURN server and the browser is chrome with latest updates.
The only thing whih I haven't done is enable TLS in kamailio and create certs. (which
I'm not completely sure how to do)..
Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message --------
From: Gonzalo Gasca Meza <gascagonzalo(a)gmail.com>
Date:27/01/2015 4:07 AM (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS communication? If yes which
version of Kamailio and rtp-proxy ?
Have you tried passing media directly between Browser and Kamailio with any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> *
media: sipml5
------------------------------------------------------------------------> *
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultimate(a)gmail.com> wrote:
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfuchs(a)sipwise.com> wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where
the SIP server sends 183 session in progress to kamailio but after that
I can only see -
"STUN: using this candidate"
"Successful STUN binding request from .."
"SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set
up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set
up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Are you running a very old OpenSSL version by any chance?
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
--
Warm Regds.
MathuRahul
_______________________________________________
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
27 Jan
27 Jan
12:21 p.m.
New subject: Need help on WebRTC with Kamailio as proxy
Any thoughts on this gents ?
On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate <rahul.ultimate(a)gmail.com>
wrote:
Kamailio is just acting as a proxy and protocol
modifier so to say. It is
workin with rtpengine from sipwise to handle media as evident from he logs.
This architectue uses a TURN server and the browser is chrome with
latest updates.
The only thing whih I haven't done is enable TLS in kamailio and create
certs. (which I'm not completely sure how to do)..
Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message --------
From: Gonzalo Gasca Meza
Date:27/01/2015 4:07 AM (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS communication?
If yes which version of Kamailio and rtp-proxy ?
Have you tried passing media directly between Browser and Kamailio with
any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip
udp--> *
media: sipml5
------------------------------------------------------------------------> *
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultimate(a)gmail.com>
wrote:
--
Warm Regds.
MathuRahul
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfuchs(a)sipwise.com>
wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
--
Warm Regds.
MathuRahul
_______________________________________________
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
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where
the SIP server sends 183 session in progress to kamailio but after that
I can only see -
"STUN: using this candidate"
"Successful STUN binding request from .."
"SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port
30794] Failed to set up SRTP after DTLS
negotiation: no SRTP protection
profile negotiated
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port
30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection
profile negotiated
Are you running a very old OpenSSL version by any chance?
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
7:31 p.m.
New subject: Need help on WebRTC with Kamailio as proxy
Hi Rahul,
Don't take me wrong, but you still have some homework to do. Apache is
not a requirement for webrtc ( apart hosting the website ). The only
difference between using http and https is that by default on http, most
browsers will always ask for the user to confirm usage of the mic/cam.
WS and WSS works with Kamailio, it's only a question of configuration (
for which there are many examples, most are broken but easily fixed, for
example for https://github.com/caruizdiaz/kamailio-ws , it's only fixing
the record routes to get sip2ws signaling working ).
Regarding rtp, you have to use rtpengine ( master from repo, not a
release, dtls broken in latest 3.7.1, fixed in 3.8 ) or something else
to be able to terminate ICE/DTLS when remote endpoints don't support
them ( most of SIP ua's today unfortunately ), again, read, experiment,
you'll eventually get it and the most important, know how your platform
works !
Start with basic browser to browser calls, without a rtp proxy, it
should work almost out of the box, then you can add some functionnality
to the basic scenario, and I'll be glad to point you to the right
direction !
Good luck !
Le 27/01/2015 03:21, Rahul MathuR a écrit :
Any thoughts on this gents ?
On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate
<rahul.ultimate(a)gmail.com <mailto:rahul.ultimate@gmail.com>> wrote:
Kamailio is just acting as a proxy and protocol modifier so to
say. It is workin with rtpengine from sipwise to handle media as
evident from he logs.
This architectue uses a TURN server and the browser is chrome
with latest updates.
The only thing whih I haven't done is enable TLS in kamailio and
create certs. (which I'm not completely sure how to do)..
Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message --------
From: Gonzalo Gasca Meza
Date:27/01/2015 4:07 AM (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS
communication? If yes which version of Kamailio and rtp-proxy ?
Have you tried passing media directly between Browser and Kamailio
with any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS
--sip udp--> *
media: sipml5
------------------------------------------------------------------------>
*
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR
<rahul.ultimate(a)gmail.com <mailto:rahul.ultimate@gmail.com>> wrote:
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs
<rfuchs(a)sipwise.com <mailto:rfuchs@sipwise.com>> wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally struck at a point while implementing
Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN
server
(rfc5766-turn-server for ICE/STUN). I am able to get
to the point where
the SIP server sends 183 session in progress to
kamailio but after that
I can only see -
"STUN: using this candidate"
"Successful STUN binding request from .."
"SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]:
[tsb1jrsqsadn33jjsi4f port 30794] Failed to set up
SRTP after DTLS negotiation: no SRTP protection
profile negotiated
Jan 27 00:35:46 localhost rtpengine[5262]:
[tsb1jrsqsadn33jjsi4f port 30794] Failed to set up
SRTP after DTLS negotiation: no SRTP protection
profile negotiated
Are you running a very old OpenSSL version by any chance?
cheers
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Warm Regds.
MathuRahul
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Warm Regds.
MathuRahul
_______________________________________________
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
3398
days inactive
3398
days old
2 comments
3 participants
participants (3)
-
Rahul MathuR -
rahul.ultimate
-
Tristan Mahé