<p>Hi Nick,</p>
<p>When I first hit the same issue where media was to be sent/received from a different IP other than signalling I used 'r' flag in the engage_rtpproxy() function. I don't think you are using this function rather using offer and answer functions which I'd say a manual approach. <br>
Tell me if using the 'r' flag in both offer and answer helps with these calls. <br>
Umm...and yeah also check with your media server not to directmedia with the carrier. </p>
<p>Thanks<br>
Sammy</p>
<div class="gmail_quote">On Mar 21, 2013 1:45 AM, "Nick Khamis" <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello Razvan, and Sammy,<br>
<br>
I really appreciate any help I can get in the matter. Not to sound<br>
like a broken record, but when the SIP and RTP stream are coming from<br>
the same source:<br>
<br>
U 2013/03/20 16:13:55.029233 <a href="http://69.147.236.82:5060" target="_blank">69.147.236.82:5060</a> -> <a href="http://192.168.2.5:5060" target="_blank">192.168.2.5:5060</a><br>
<br>
o=root 19098 19098 IN IP4 69.147.236.82.<br>
c=IN IP4 69.147.236.82.<br>
<br>
RTP and RTCP stats are flowing as expected, and we have two way audio:<br>
<br>
INFO:remove_session: RTP stats: 751 in from callee, 730 in from<br>
caller, 1481 relayed, 0 dropped<br>
INFO:remove_session: RTCP stats: 1 in from callee, 0 in from caller, 1<br>
relayed, 0 dropped<br>
<br>
<br>
It would seem that, the problem arises when the SIP and RTP streams<br>
are coming from different sources:<br>
<br>
U 2013/03/20 16:27:36.758048 <a href="http://81.201.86.45:5060" target="_blank">81.201.86.45:5060</a> -> <a href="http://192.168.2.5:5060" target="_blank">192.168.2.5:5060</a><br>
o=root 5539 5539 IN IP4 81.201.86.26.<br>
c=IN IP4 81.201.86.26<br>
<br>
RTPProxy reports that RTP traffic is flowing from the callee, but in<br>
fact there is no audio both ways:<br>
<br>
INFO:remove_session: RTP stats: 148 in from callee, 0 in from caller,<br>
148 relayed, 0 dropped<br>
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0<br>
relayed, 0 dropped<br>
<br>
Commenting out the RTP stuff in OpenSIPS leads to one way outgoing<br>
audio. Looking closer at the RTP proxy log I see that it's prefilling<br>
the caller's address to the source of the SIP messages and not to the<br>
source of the RTP stream:<br>
<br>
INFO:handle_command: pre-filling caller's address with <a href="http://81.201.86.45:10118" target="_blank">81.201.86.45:10118</a><br>
<br>
A trace on the RTP stream confirms this:<br>
<br>
130.427203 192.168.2.100 -> 192.168.2.5 UDP 214 Source port: vtsas<br>
Destination port: 9624<br>
130.427338 192.168.2.5 -> 81.201.86.45 UDP 214 Source port: 30060<br>
Destination port: 17020<br>
130.468931 192.168.2.100 -> 192.168.2.5 UDP 214 Source port: vtsas<br>
Destination port: 9624<br>
130.468985 192.168.2.5 -> 81.201.86.45 UDP 214 Source port: 30060<br>
Destination port: 17020<br>
<br>
Thanks in Advance,<br>
<br>
Nick.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
On 3/19/13, Nick Khamis <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
> Hello Razvan,<br>
><br>
> I should have mentioned that we only experienced this issue with this<br>
> particular DID provider. With others everything works perfectly. We<br>
> suspect the issue is because the RTP stream is coming from a different<br>
> source that of the SIP messages. So I think it's a matter of lining up<br>
> rtpproxy_offer/answer parameters (i.e., co).<br>
><br>
> Unfortunately, their service to our zone today is down. Will post<br>
> detailed logs as soon as we can initiate some calls.<br>
><br>
> Nick.<br>
><br>
> On 3/19/13, Răzvan Crainea <<a href="mailto:razvan@opensips.org">razvan@opensips.org</a>> wrote:<br>
>> Hi, Nick!<br>
>><br>
>> You said that you can see logs for RTPProxy. Can you set the debug level<br>
>> to DBUG and paste (preferably on pastebin) the logs of the session?<br>
>><br>
>> Best regards,<br>
>><br>
>> Razvan Crainea<br>
>> OpenSIPS Core Developer<br>
>> <a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a><br>
>><br>
>> On 03/19/2013 03:52 PM, Nick Khamis wrote:<br>
>>> I wanted to mention that the same setup works perfectly with VoIP.ms<br>
>>> but not Voxbone. I think the problem is that the SIP messages and RTP<br>
>>> stream for voxbone are coming from different sources. With other<br>
>>> origination providers SIP and RTP streams came from the same source,<br>
>>> so we never experienced a problem.<br>
>>><br>
>>> We are currently looking into rtpproxy_orffer/answer parameters (i..e,<br>
>>> "reico"...) to see if we can line things up nicely.<br>
>>><br>
>>> Nichola.<br>
>>><br>
>>> On 3/19/13, Nick Khamis <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
>>>> RTPProxy does work behind NAT. It's mediaporxy that must be on a public<br>
>>>> ip.<br>
>>>><br>
>>>> Thanks for your help.<br>
>>>><br>
>>>> Nick.<br>
>>>><br>
>>>> On 3/19/13, Muhammad Shahzad <<a href="mailto:shaheryarkh@gmail.com">shaheryarkh@gmail.com</a>> wrote:<br>
>>>>> If you are unfamiliar with rtp proxy and how it works, then it would<br>
>>>>> be<br>
>>>>> better for you to use engage_rtp_proxy rather then offer / answer<br>
>>>>> model.<br>
>>>>> Also RTP Proxy requires public IP address, its likely not to work on<br>
>>>>> private subnets (unless you have all SIP entities on same LAN, in<br>
>>>>> which<br>
>>>>> case theoretically it should work but i have never tested it myself).<br>
>>>>><br>
>>>>> Thank you.<br>
>>>>><br>
>>>>><br>
>>>>> On Mon, Mar 18, 2013 at 4:18 PM, Nick Khamis <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
>>>>><br>
>>>>>> 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<br>
>>>>>> 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<br>
>>>>>> <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<br>
>>>>>> <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,<br>
>>>>>> 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,<br>
>>>>>> 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<br>
>>>>>> network?<br>
>>>>>><br>
>>>>>> Thanks in Advance,<br>
>>>>>><br>
>>>>>> Nick.<br>
>>>>>><br>
>>>>>> _______________________________________________<br>
>>>>>> Users mailing list<br>
>>>>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>>>>>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>>>>>><br>
>>>>><br>
>>>>><br>
>>>>> --<br>
>>>>> Mit freundlichen Grüßen<br>
>>>>> Muhammad Shahzad<br>
>>>>> -----------------------------------<br>
>>>>> CISCO Rich Media Communication Specialist (CRMCS)<br>
>>>>> CISCO Certified Network Associate (CCNA)<br>
>>>>> Cell: <a href="tel:%2B49%20176%2099%2083%2010%2085" value="+4917699831085">+49 176 99 83 10 85</a><br>
>>>>> MSN: <a href="mailto:shari_786pk@hotmail.com">shari_786pk@hotmail.com</a><br>
>>>>> Email: <a href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a><br>
>>>>><br>
>>> _______________________________________________<br>
>>> Users mailing list<br>
>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>><br>
>><br>
>> _______________________________________________<br>
>> Users mailing list<br>
>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>><br>
><br>
</blockquote></div>