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...
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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
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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Warm Regds. MathuRahul
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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Gonzalo Gasca Meza -
Rahul MathuR -
Richard Fuchs -
Tristan Mahé