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

[Răzvan Crainea]

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 <symack at 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 <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
>>>
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