Ok, that brings some clarification. Thanks.<br><br>So the correct scenario is:<br>1) softphone --registered to--> opensips A (pure)<br>2) call is relayed from opensips A to opensips B (the B2B one)<br>3) the opensips B connects to the termination<br>
4) the RTP goes between the softphone -> opensips A -> rtpproxy<br><br>Would that work ? :)<br><br><br><br>
<br><br><div class="gmail_quote">On 2 February 2011 20:04, Ovidiu Sas <span dir="ltr"><<a href="mailto:osas@voipembedded.com">osas@voipembedded.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
The nathelper module is performing changes on the received INVITE<br>
(changing the SDP).<br>
Those changes are not visible by the b2b module and therefor discarded.<br>
As a result, the nathelper module (and any module that is changing the<br>
initial INVITE) doesn't work with the b2b module.<br>
The only change visible to the b2b module is the RURI.<br>
<br>
Regards,<br>
<font color="#888888">Ovidiu Sas<br>
</font><div><div></div><div class="h5"><br>
On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov <<a href="mailto:kamen.petrov@gmail.com">kamen.petrov@gmail.com</a>> wrote:<br>
> Hi Ovidu,<br>
><br>
> I do not perform any changes on the received invite.<br>
><br>
> The "top hiding" does it and the problem is.. it does not change only the<br>
> media IP. Everything else goes OK.<br>
><br>
> Are you saying the "top hiding" does not work properly with the nathelper ?<br>
><br>
> Thanks<br>
> -- Kamen<br>
><br>
><br>
><br>
><br>
> On 2 February 2011 19:41, Ovidiu Sas <<a href="mailto:osas@voipembedded.com">osas@voipembedded.com</a>> wrote:<br>
>><br>
>> The B2B module is operating on the received INVITE. Any changes that<br>
>> you make to the received INVITE are not visible by the B2B module.<br>
>> Use a proxy to perform whatever you want to do (rtpproxy, accounting,<br>
>> etc.) and a separate server only for b2b (top hiding).<br>
>><br>
>> Regards,<br>
>> Ovidiu Sas<br>
>><br>
>> On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov <<a href="mailto:kamen.petrov@gmail.com">kamen.petrov@gmail.com</a>><br>
>> wrote:<br>
>> > Hi Guys,<br>
>> ><br>
>> > I am testing the following call flow:<br>
>> > Soft Phone => opensips (configured for B2B) => third party termination<br>
>> > SIP<br>
>> > proxy<br>
>> ><br>
>> > Here is my config:<br>
>> ><br>
>> > modparam("b2b_entities", "script_req_route", "b2b_request")<br>
>> > modparam("b2b_entities", "script_reply_route", "b2b_reply")<br>
>> ><br>
>> ><br>
>> ><br>
>> > local_route {<br>
>> > xlog("================LOCAL_ROUTE ($rm - $rr)============\n");<br>
>> > setflag(22);<br>
>> > if (is_method("INVITE")) {<br>
>> > engage_rtp_proxy("e","<OPENSIPS_IP>");<br>
>> > exit;<br>
>> > }<br>
>> > else if (is_method("BYE") ) {<br>
>> > xlog("================BYE============\n");<br>
>> > }<br>
>> > }<br>
>> ><br>
>> ><br>
>> > route[b2b_request] {<br>
>> > $avp(s:source_ip_address) := $si;<br>
>> > perl_exec("messagedump_route", "messages");<br>
>> > xlog("b2b_request ($ci) ($rm - $rr)\n");<br>
>> > }<br>
>> ><br>
>> ><br>
>> > route[b2b_reply] {<br>
>> > $avp(s:source_ip_address) := $si;<br>
>> > perl_exec("messagedump_reply", "messages");<br>
>> > xlog("b2b_reply ($ci) - $rm - $rr\n");<br>
>> > }<br>
>> ><br>
>> ><br>
>> > route{<br>
>> > ...<br>
>> > if (is_method("INVITE") &&<br>
>> > perl_exec("check_for_forwarding_number"))<br>
>> > {<br>
>> > engage_rtp_proxy("e","<OPENSIPS_IP>");<br>
>> > xlog("LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD<br>
>> > TO:<br>
>> > $avp(s:fwd_ip)\n");<br>
>> > setflag(1); # do accounting<br>
>> > xlog("L_ERR", "LOG: to uri=[$tu]<br>
>> > [$avp(s:sip_proxy_ip)]\n");<br>
>> ><br>
>> > b2b_init_request("top hiding");<br>
>> > exit;<br>
>> > };<br>
>> ><br>
>> > ...<br>
>> > }<br>
>> ><br>
>> ><br>
>> > What happens is:<br>
>> > - INVITE from the soft phone to the opensips<br>
>> > - catched by the B2B and relayed to the third party SIP proxy + trying<br>
>> > returned to the soft phone<br>
>> > - "Session Progress" received from the third party SIP proxy -> opensips<br>
>> > -><br>
>> > my soft phone<br>
>> ><br>
>> > At that stage, here is what I have on the soft phone log:<br>
>> > 18:56:50 UDP Packet Received from <OPENSIPS_IP>:5060<br>
>> > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<br>
>> > SIP/2.0 183 Session Progress<br>
>> > Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604<br>
>> > To: <sip:359883409291@<OPENSIPS_DOMAIN>:5060>;tag=B2B.113.667<br>
>> > From: "359883327749" <sip:359883327749@<OPENSIPS_DOMAIN>:5060>;tag=1040<br>
>> > Call-ID: <a href="mailto:1296636915-3604-SALASWORK@192.168.1.2">1296636915-3604-SALASWORK@192.168.1.2</a><br>
>> > CSeq: 361 INVITE<br>
>> > Content-Type: application/sdp<br>
>> > Contact: <sip:<OPENSIPS_IP>:5060;transport=udp><br>
>> > Server: OpenSIPS (1.6.3-notls (x86_64/linux))<br>
>> > Content-Length: 184<br>
>> ><br>
>> > v=0<br>
>> > o=SBCSIPUAS 900116523 1 IN IP4 <THIRD_PARTY_SIP_PROXY_IP><br>
>> > s=SBCSIPUAS SIP STACK v1.0<br>
>> > c=IN IP4 <THIRD_PARTY_SIP_PROXY_IP><br>
>> > t=0 0<br>
>> > m=audio 17900 RTP/AVP 0<br>
>> > a=rtpmap:0 PCMU/8000<br>
>> > a=sendrecv<br>
>> > a=maxptime:30<br>
>> ><br>
>> ><br>
>> > As can be seen, the media IP is not rewritten by the opensips and the IP<br>
>> > passed to my soft phone is the IP of the termination IP for the opensips<br>
>> > (i.e. the third party SIP proxy IP). Because of that, my soft phone<br>
>> > starts<br>
>> > the RTP directly to my provider instead trough the RTP proxy that is<br>
>> > attached to the opensips.<br>
>> > Just to clarify, the media IP of my soft phone is not passed to my<br>
>> > provider<br>
>> > - that case is handled good.<br>
>> ><br>
>> ><br>
>> > Any idea what is missing ?<br>
>> ><br>
>> > Thanks in advance.<br>
>> > -- Kamen<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>
>> 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>
<br>
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</div></div></blockquote></div><br>