26 Jan
2015
26 Jan
'15
8:21 p.m.
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"
a) What should I do to resolve this issue ?
b) Why is that I never get 200 OK for the INVITE ?
I am attaching the logfile & configs herewith.
Can somebody please help me out here...
--
Warm Regds.
MathuRahul
26 Jan
26 Jan
8:43 p.m.
Hi Rahul,
Check your record-route, it should not be static. ( maybe use
advertised_address and allow double record-route for rr module ).
My 2 cents.
Le 26/01/2015 11:21, Rahul MathuR a écrit :
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"
a) What should I do to resolve this issue ?
b) Why is that I never get 200 OK for the INVITE ?
I am attaching the logfile & configs herewith.
Can somebody please help me out here...
--
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
9:30 p.m.
Hi Tristan,
Thanks for looking into it.
I toggled advertised_address to static one, but no luck :(
On Tue, Jan 27, 2015 at 1:13 AM, Tristan Mahé <t.mahe(a)b-and-c.net> wrote:
Hi Rahul,
Check your record-route, it should not be static. ( maybe use
advertised_address and allow double record-route for rr module ).
My 2 cents.
Le 26/01/2015 11:21, Rahul MathuR a écrit :
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"
a) What should I do to resolve this issue ?
b) Why is that I never get 200 OK for the INVITE ?
I am attaching the logfile & configs herewith.
Can somebody please help me out here...
--
Warm Regds.
MathuRahul
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://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
--
Warm Regds.
MathuRahul
9:34 p.m.
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
9:44 p.m.
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
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
9:59 p.m.
Hello Richard,
Now, after upgrading to - OpenSSL 1.0.1j 15 Oct 2014
Errors like -
"Failed to set up SRTP after DTLS negotiation: no SRTP protection profile
negotiated" are not seen but still the call is not getting through.
Please let me know how to proceed..
Thanks in advance
On Tue, Jan 27, 2015 at 2:14 AM, 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:
--
Warm Regds.
MathuRahul
On 26/01/15 02:21 PM, Rahul MathuR wrote:
--
Warm Regds.
MathuRahul
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
11:37 p.m.
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:
--
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
3596
days inactive
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6 comments
4 participants
participants (4)
-
Gonzalo Gasca Meza -
Rahul MathuR -
Richard Fuchs -
Tristan Mahé