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 <abalas...@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
--
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
