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<tt>Hi, Vladimir!<br>
<br>
Please provide us a complete SIP trace of the call. You can send
me on private if you want.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">--
Răzvan Crainea
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 04/20/2012 05:05 PM, Vladimir Vakulov wrote:
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<p class="MsoNormal" style="background:whitesmoke"><span
style="font-size:12.0pt;font-family:"Times New
Roman","serif"" lang="EN">Good afternoon.<br>
<br>
For several days I try to construct a scheme LB RTPProxy:<br>
incoming calls -> GW [10.2.42.70] -> [10.2.42.74]
OpenSIPs + RTPProxy [192.168.1.2] -> [192.168.1.3 (/
4/5/6)] 4x Asterisks (in a test using only the 192.168.1.3)<br>
Calls come in Asterisk, but RTP session is
established directly between Asterisk andGW. When I add the
script to use RTPProxy, it appears only in
the INVITE message and RTP session is established
between 192.168.1.2 and 192.168.1.3, but not with
thecallers. In the "ACK" and "200 OK" everything as
it was before.<br>
Example of 200<br>
"<br>
<--- SIP read from UDP: 10.2.42.74:5060 ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 10.2.42.70:5060; received = 10.2.42.70;
branch = z9hG4bK342cdd3c; rport = 5060<br>
Record-Route:
<sip:192.168.1.2;r2=on;lr;ftag=as3d53f405;did=0e8.e8c24272><br>
Record-Route:
<sip:10.2.42.74;r2=on;lr;ftag=as3d53f405;did=0e8.e8c24272><br>
From: "New User" <a class="moz-txt-link-rfc2396E" href="mailto:sip:12345@10.2.42.70"><sip:12345@10.2.42.70></a>; tag =
as3d53f405<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:7441@10.2.42.74:5060"><sip:7441@10.2.42.74:5060></a>; tag = as3b390137<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:6e15f78d6bcdbb293620488a251174f8@10.2.42.70">6e15f78d6bcdbb293620488a251174f8@10.2.42.70</a>: 5060<br>
CSeq: 102 INVITE<br>
Server: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:7441@192.168.1.3:5060"><sip:7441@192.168.1.3:5060></a><br>
Content-Type: application / sdp<br>
Content-Length: 255<br>
<br>
v = 0,<br>
o = root 1604383785 1604383785 IN IP4 192.168.1.3<br>
s = Asterisk PBX 10.3.0<br>
c = IN IP4 192.168.1.3<br>
t = 0 0<br>
m = audio 18100 RTP / AVP 8 0 3111<br>
a = rtpmap: 8 PCMA/8000<br>
a = rtpmap: 0 PCMU/8000<br>
a = rtpmap: 3 GSM/8000<br>
a = rtpmap: 111 G726-32/8000<br>
a = ptime: 20<br>
a = sendrecv<br>
"<br>
<br>
I would be grateful for any help.</span><span
style="font-size:10.0pt;font-family:"Times New
Roman","serif"" lang="EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Times New
Roman","serif"" lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Times New
Roman","serif"" lang="EN-US">Vladimir.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-family:"Times New
Roman","serif"" lang="EN-US"><o:p> </o:p></span></p>
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