[OpenSIPS-Users] RTPProxy Support - Not prefilling callees address

[Muhammad Shahzad]

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 <symack at 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 at 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 at 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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> Users at lists.opensips.org
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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_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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