If you are unfamiliar with rtp proxy and how it works, then it would be better for you to use engage_rtp_proxy rather then offer / answer model. Also RTP Proxy requires public IP address, its likely not to work on private subnets (unless you have all SIP entities on same LAN, in which case theoretically it should work but i have never tested it myself).<div>
<br></div><div>Thank you.</div><div><br><br><div class="gmail_quote">On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com" target="_blank">symack@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I am not sure if this is the correct place to post OpenSIPS+RTPProxy<br>
questions however, I tried to subscribing to the RTP proxy mailing<br>
list and never heard from them since. If it is ok to post RTP proxy<br>
related questions here.... I am trying to test OpenSIPS with RTP proxy<br>
with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,<br>
RTPPoxy) just for testing.<br>
<br>
The code I am using is:<br>
<br>
route {<br>
force_rport();<br>
}<br>
route[1] {<br>
if (is_method("INVITE")) {<br>
t_on_branch("1");<br>
t_on_reply("1");<br>
t_on_failure("1");<br>
<br>
if (has_body("application/sdp")) rtpproxy_offer();<br>
}<br>
else if (is_method("BYE|CANCEL")) {<br>
unforce_rtp_proxy();<br>
}<br>
<br>
if (!t_relay()) {<br>
sl_reply_error();<br>
};<br>
exit;<br>
}<br>
onreply_route[1] {<br>
if (has_body("application/sdp")) rtpproxy_answer();<br>
}<br>
<br>
<br>
There is no way audio using RTP proxy, but audio is fine between the<br>
UA without including the RTP proxy related script. Looking at the log<br>
I found that RTP is prefilling the callers address twice, but not the<br>
callees address.<br>
<br>
<br>
INFO:main: rtpproxy started, pid 7287<br>
INFO:handle_command: new session<br>
ae450168-538e-e211-8550-001b7700a65b@oakville, tag<br>
d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong<br>
INFO:handle_command: new session on a port 35010 created, tag<br>
d23f0168-538e-e211-8550-001b7700a65b;1<br>
INFO:handle_command: pre-filling caller's address with <a href="http://192.168.2.101:5062" target="_blank">192.168.2.101:5062</a><br>
INFO:handle_command: new session<br>
ae450168-538e-e211-8550-001b7700a65b@oakville, tag<br>
d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong<br>
INFO:handle_command: new session on a port 22982 created, tag<br>
d23f0168-538e-e211-8550-001b7700a65b;2<br>
INFO:handle_command: pre-filling caller's address with <a href="http://192.168.2.101:5064" target="_blank">192.168.2.101:5064</a><br>
INFO:handle_delete: forcefully deleting session 1 on ports 35010/0<br>
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0<br>
relayed, 0 dropped<br>
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0<br>
relayed, 0 dropped<br>
INFO:remove_session: session on ports 35010/0 is cleaned up<br>
INFO:handle_delete: forcefully deleting session 2 on ports 22982/0<br>
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0<br>
relayed, 0 dropped<br>
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0<br>
relayed, 0 dropped<br>
INFO:remove_session: session on ports 22982/0 is cleaned up<br>
<br>
Is it possible to test RTP relaying with everything on the same network?<br>
<br>
Thanks in Advance,<br>
<br>
Nick.<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div><span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Mit freundlichen Grüßen</span></div><span style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">Muhammad Shahzad</span><br style="color:rgb(136,136,136);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
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