hi
I have configured kamailio and freeswitch on two Ubuntu servers as follows.
Sip_TRUNK------> kamailio-------> freeswitch
i wan to manage rtp through the kamailio server,
first i want to know,
1. can i do it without RTP Proxy ?
anyway i tried it with RTP Proxy as follows,
I installed rtpproxy on kamailio server and start it using
“rtpproxy –F -l pu.bl.ic.ip -s udp:localhost :7722 -u root” it was working
Then I put
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")
modparam("rtpproxy", "rtpproxy_retr", 2)
modparam("rtpproxy", "ice_candidate_priority_avp", "$avp(ice_priority)")
mhomed=0
route[FROM_TRUNK]
force_send_socket("pu.bl.ic.ip ");
rtpproxy_manage("","pu.bl.ic.ip ");
$du="sip:ip.of.free.switch:5080";
route(RELAY);
then call is received to user and disconnected
on freeswitch console get some errors
[ERR] switch_core_media.c:7179 AUDIO RTP REPORTS ERROR: [Remote Address Error!] [ERR] mod_sofia.c:2290 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3679035755 3679035755 IN IP4 pu.bl.ic.ippu.bl.ic.ip s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4167841678 RTP/AVP 123 8 0 101 c=IN IP4 pu.bl.ic.ippu.bl.ic.ip b=TIAS:64000 a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=16000;useinbandfec=1 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp:4167941679 a=nortpproxy:yes a=nortpproxy:yes
please advice me thank you
As long as the media endpoint on the "Sip_TRUNK" side has direct network and transport-layer reachability to the Freeswitch, there should be no need for an intervening rtpproxy, and in fact, life would be much simpler without it.
This would require both the service provider side and the Freeswitch to have public IP addresses with no firewall rules in between that would prevent the free flow of RTP between them across a broad range of UDP ports. If one or both endpoints are behind NAT and/or on a private network, you're going to need an rtpproxy between them.
-- Alex