Hi, Nick!
You said that you can see logs for RTPProxy. Can you set the debug level
to DBUG and paste (preferably on pastebin) the logs of the session?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/19/2013 03:52 PM, Nick Khamis wrote:
I wanted to mention that the same setup works perfectly with VoIP.ms
but not Voxbone. I think the problem is that the SIP messages and RTP
stream for voxbone are coming from different sources. With other
origination providers SIP and RTP streams came from the same source,
so we never experienced a problem.
We are currently looking into rtpproxy_orffer/answer parameters (i..e,
"reico"...) to see if we can line things up nicely.
Nichola.
On 3/19/13, Nick Khamis <sym...@gmail.com> wrote:
RTPProxy does work behind NAT. It's mediaporxy that must be on a public ip.
Thanks for your help.
Nick.
On 3/19/13, Muhammad Shahzad <shaherya...@gmail.com> wrote:
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).
Thank you.
On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis <sym...@gmail.com> wrote:
I am not sure if this is the correct place to post OpenSIPS+RTPProxy
questions however, I tried to subscribing to the RTP proxy mailing
list and never heard from them since. If it is ok to post RTP proxy
related questions here.... I am trying to test OpenSIPS with RTP proxy
with everything behind the same NAT box (i.e., 2 UAs, OpenSIPS,
RTPPoxy) just for testing.
The code I am using is:
route {
force_rport();
}
route[1] {
if (is_method("INVITE")) {
t_on_branch("1");
t_on_reply("1");
t_on_failure("1");
if (has_body("application/sdp")) rtpproxy_offer();
}
else if (is_method("BYE|CANCEL")) {
unforce_rtp_proxy();
}
if (!t_relay()) {
sl_reply_error();
};
exit;
}
onreply_route[1] {
if (has_body("application/sdp")) rtpproxy_answer();
}
There is no way audio using RTP proxy, but audio is fine between the
UA without including the RTP proxy related script. Looking at the log
I found that RTP is prefilling the callers address twice, but not the
callees address.
INFO:main: rtpproxy started, pid 7287
INFO:handle_command: new session
ae450168-538e-e211-8550-001b7700a65b@oakville, tag
d23f0168-538e-e211-8550-001b7700a65b;1 requested, type strong
INFO:handle_command: new session on a port 35010 created, tag
d23f0168-538e-e211-8550-001b7700a65b;1
INFO:handle_command: pre-filling caller's address with
192.168.2.101:5062
INFO:handle_command: new session
ae450168-538e-e211-8550-001b7700a65b@oakville, tag
d23f0168-538e-e211-8550-001b7700a65b;2 requested, type strong
INFO:handle_command: new session on a port 22982 created, tag
d23f0168-538e-e211-8550-001b7700a65b;2
INFO:handle_command: pre-filling caller's address with
192.168.2.101:5064
INFO:handle_delete: forcefully deleting session 1 on ports 35010/0
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 35010/0 is cleaned up
INFO:handle_delete: forcefully deleting session 2 on ports 22982/0
INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 22982/0 is cleaned up
Is it possible to test RTP relaying with everything on the same network?
Thanks in Advance,
Nick.
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--
Mit freundlichen Grüßen
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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