Hi Arjun,
If you have no audio at all for the call to Conf Server, you need to
check the signaling, particularity speaking the SDP part to be sure you
inserted RTPProxy correctly between UAC and Conf Server.
So, make a SIP capture for the call to Conf and check if the IPs in SDP
are correct.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06/07/2012 05:46 PM, Arjun Shankar K S wrote:
Hi All,
Greetings to everyone !!!
I have set up opensips and RTP Proxy in two different hosts since I
have opensips in a Natted environment where RTP Proxy refused to budge.
Now I have installed RTP Proxy in a direct public IP. Normal calls
between 2 Client is working great !!
During conference call, the calls get connected but there is no voice
communication between any of them and soon the client who was
connected last, gets disconnected.
I could not find much support regarding this issue. Any support is
sincerely appreciated.
In my opensips.cfg, I have made the following config changes for RTP
in different host,
------Nat Params-------------
modparam("usrloc","nat_bflag", 6)
modparam("nathelper","rtpproxy_sock", "udp:rtp_proxy_publicIP:7890")
modparam("nathelper","natping_interval", 30)
modparam("nathelper","ping_nated_only", 0)
modparam("nathelper","sipping_bflag", 7)
modparam("nathelper","sipping_from", "sip:pinger@PROXY_IP")
modparam("registrar","received_avp", "$avp(i:42)")
modparam("nathelper","received_avp", "$avp(i:42)")
I am running my RTP Proxy using the following command,
./rtpproxy -l rtp_proxy_publicIP-s udp:*:7890 -F
Thanks,
Arjun
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