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

[Nick Khamis]

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 <shaheryarkh at 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 <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.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> --
> 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
>