hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know. thanks.
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know.
On calls where you do not want rtpproxy to relay media, just don't use it (don't call force_rtp_proxy())?
Dear Alex Balashov, thanks for helping i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no stream transfer. the call can make but can not hear anything .
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
________________________________ From: Alex Balashov abalashov@evaristesys.com To: sr-users@lists.sip-router.org Sent: Tue, August 24, 2010 4:58:27 PM Subject: Re: [SR-Users] help to configure RTP stream with NAT.
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know.
On calls where you do not want rtpproxy to relay media, just don't use it (don't call force_rtp_proxy())?
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
_______________________________________________ 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
Without rtpproxy or mediaproxy, the both SIP clients have to be reached from Internet, or it has to have the public IP.
But in your case, I don't think you can have both client on Internet.
Tung
From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of truong ngoc THANH Sent: Tuesday, August 24, 2010 5:24 PM To: Alex Balashov Cc: kamailio Subject: Re: [SR-Users] help to configure RTP stream with NAT.
Dear Alex Balashov, thanks for helping i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no stream transfer. the call can make but can not hear anything .
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
_____
From: Alex Balashov abalashov@evaristesys.com To: sr-users@lists.sip-router.org Sent: Tue, August 24, 2010 4:58:27 PM Subject: Re: [SR-Users] help to configure RTP stream with NAT.
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know.
On calls where you do not want rtpproxy to relay media, just don't use it (don't call force_rtp_proxy())?
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
_______________________________________________ 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
In that case, there is a network or transport-layer reachability issue between the two clients.
On 08/24/2010 06:24 AM, truong ngoc THANH wrote:
Dear Alex Balashov, thanks for helping i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no stream transfer. the call can make but can not hear anything .
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
*From:* Alex Balashov abalashov@evaristesys.com *To:* sr-users@lists.sip-router.org *Sent:* Tue, August 24, 2010 4:58:27 PM *Subject:* Re: [SR-Users] help to configure RTP stream with NAT.
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know.
On calls where you do not want rtpproxy to relay media, just don't use it (don't call force_rtp_proxy())?
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Alex Balashov, two clients is behind NAT, when i configure nathelper, the call make ok, but RTP proxy handle media stream, I want to make media stream go direct from sip client to another. so is there any solve ?
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
________________________________ From: Alex Balashov abalashov@evaristesys.com To: truong ngoc THANH ngoc217thanh@yahoo.com Cc: kamailio sr-users@lists.sip-router.org Sent: Tue, August 24, 2010 5:41:34 PM Subject: Re: [SR-Users] help to configure RTP stream with NAT.
In that case, there is a network or transport-layer reachability issue between the two clients.
On 08/24/2010 06:24 AM, truong ngoc THANH wrote:
Dear Alex Balashov, thanks for helping i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no stream transfer. the call can make but can not hear anything .
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
*From:* Alex Balashov abalashov@evaristesys.com *To:* sr-users@lists.sip-router.org *Sent:* Tue, August 24, 2010 4:58:27 PM *Subject:* Re: [SR-Users] help to configure RTP stream with NAT.
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know.
On calls where you do not want rtpproxy to relay media, just don't use it (don't call force_rtp_proxy())?
-- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
2010/8/24 truong ngoc THANH ngoc217thanh@yahoo.com:
Hi Alex Balashov, two clients is behind NAT,
Same NAT or different NAT?
when i configure nathelper, the call make ok, but RTP proxy handle media stream, I want to make media stream go direct from sip client to another. so is there any solve ?
STUN / ICE in both endpoints. If not, you need a RTP tunnel solution (as RtpProxy). Magic doesn't exist.
hi, my diagram is : sip client <--> nat <--> kamailio server <--> nat <--> sip client (something like that). you said that i can use "RTP tunnel solution"(as RtpProxy ?). I hope that kamailio handle signaling protocol and media stream go direct from sip client to another . thanks for help.
TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh
________________________________ From: Iñaki Baz Castillo ibc@aliax.net To: truong ngoc THANH ngoc217thanh@yahoo.com Cc: Alex Balashov abalashov@evaristesys.com; kamailio sr-users@lists.sip-router.org Sent: Thu, August 26, 2010 5:02:37 AM Subject: Re: [SR-Users] help to configure RTP stream with NAT.
2010/8/24 truong ngoc THANH ngoc217thanh@yahoo.com:
Hi Alex Balashov, two clients is behind NAT,
Same NAT or different NAT?
when i configure nathelper, the call make ok, but RTP proxy handle media stream, I want to make media stream go direct from sip client to another. so is there any solve ?
STUN / ICE in both endpoints. If not, you need a RTP tunnel solution (as RtpProxy). Magic doesn't exist.
2010/8/26 truong ngoc THANH ngoc217thanh@yahoo.com:
hi, my diagram is : sip client <--> nat <--> kamailio server <--> nat <--> sip client (something like that). you said that i can use "RTP tunnel solution"(as RtpProxy ?). I hope that kamailio handle signaling protocol and media stream go direct from sip client to another . thanks for help.
Again:
STUN / ICE in both endpoints. If not, you need a RTP tunnel solution (as RtpProxy). Magic doesn't exist.
-
Alex Balashov -
Iñaki Baz Castillo -
truong ngoc THANH -
Tung Tran