[OpenSIPS-Users] B2B + top hiding + origination media IP
Ovidiu Sas
osas at voipembedded.com
Wed Feb 2 19:16:12 CET 2011
yes!
On Wed, Feb 2, 2011 at 1:12 PM, Kamen Petrov <kamen.petrov at gmail.com> wrote:
> Ok, that brings some clarification. Thanks.
>
> So the correct scenario is:
> 1) softphone --registered to--> opensips A (pure)
> 2) call is relayed from opensips A to opensips B (the B2B one)
> 3) the opensips B connects to the termination
> 4) the RTP goes between the softphone -> opensips A -> rtpproxy
>
> Would that work ? :)
>
>
>
>
>
> On 2 February 2011 20:04, Ovidiu Sas <osas at voipembedded.com> wrote:
>>
>> The nathelper module is performing changes on the received INVITE
>> (changing the SDP).
>> Those changes are not visible by the b2b module and therefor discarded.
>> As a result, the nathelper module (and any module that is changing the
>> initial INVITE) doesn't work with the b2b module.
>> The only change visible to the b2b module is the RURI.
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov <kamen.petrov at gmail.com>
>> wrote:
>> > Hi Ovidu,
>> >
>> > I do not perform any changes on the received invite.
>> >
>> > The "top hiding" does it and the problem is.. it does not change only
>> > the
>> > media IP. Everything else goes OK.
>> >
>> > Are you saying the "top hiding" does not work properly with the
>> > nathelper ?
>> >
>> > Thanks
>> > -- Kamen
>> >
>> >
>> >
>> >
>> > On 2 February 2011 19:41, Ovidiu Sas <osas at voipembedded.com> wrote:
>> >>
>> >> The B2B module is operating on the received INVITE. Any changes that
>> >> you make to the received INVITE are not visible by the B2B module.
>> >> Use a proxy to perform whatever you want to do (rtpproxy, accounting,
>> >> etc.) and a separate server only for b2b (top hiding).
>> >>
>> >> Regards,
>> >> Ovidiu Sas
>> >>
>> >> On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov <kamen.petrov at gmail.com>
>> >> wrote:
>> >> > Hi Guys,
>> >> >
>> >> > I am testing the following call flow:
>> >> > Soft Phone => opensips (configured for B2B) => third party
>> >> > termination
>> >> > SIP
>> >> > proxy
>> >> >
>> >> > Here is my config:
>> >> >
>> >> > modparam("b2b_entities", "script_req_route", "b2b_request")
>> >> > modparam("b2b_entities", "script_reply_route", "b2b_reply")
>> >> >
>> >> >
>> >> >
>> >> > local_route {
>> >> > xlog("================LOCAL_ROUTE ($rm -
>> >> > $rr)============\n");
>> >> > setflag(22);
>> >> > if (is_method("INVITE")) {
>> >> > engage_rtp_proxy("e","<OPENSIPS_IP>");
>> >> > exit;
>> >> > }
>> >> > else if (is_method("BYE") ) {
>> >> > xlog("================BYE============\n");
>> >> > }
>> >> > }
>> >> >
>> >> >
>> >> > route[b2b_request] {
>> >> > $avp(s:source_ip_address) := $si;
>> >> > perl_exec("messagedump_route", "messages");
>> >> > xlog("b2b_request ($ci) ($rm - $rr)\n");
>> >> > }
>> >> >
>> >> >
>> >> > route[b2b_reply] {
>> >> > $avp(s:source_ip_address) := $si;
>> >> > perl_exec("messagedump_reply", "messages");
>> >> > xlog("b2b_reply ($ci) - $rm - $rr\n");
>> >> > }
>> >> >
>> >> >
>> >> > route{
>> >> > ...
>> >> > if (is_method("INVITE") &&
>> >> > perl_exec("check_for_forwarding_number"))
>> >> > {
>> >> > engage_rtp_proxy("e","<OPENSIPS_IP>");
>> >> > xlog("LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD
>> >> > TO:
>> >> > $avp(s:fwd_ip)\n");
>> >> > setflag(1); # do accounting
>> >> > xlog("L_ERR", "LOG: to uri=[$tu]
>> >> > [$avp(s:sip_proxy_ip)]\n");
>> >> >
>> >> > b2b_init_request("top hiding");
>> >> > exit;
>> >> > };
>> >> >
>> >> > ...
>> >> > }
>> >> >
>> >> >
>> >> > What happens is:
>> >> > - INVITE from the soft phone to the opensips
>> >> > - catched by the B2B and relayed to the third party SIP proxy +
>> >> > trying
>> >> > returned to the soft phone
>> >> > - "Session Progress" received from the third party SIP proxy ->
>> >> > opensips
>> >> > ->
>> >> > my soft phone
>> >> >
>> >> > At that stage, here is what I have on the soft phone log:
>> >> > 18:56:50 UDP Packet Received from <OPENSIPS_IP>:5060
>> >> > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>> >> > SIP/2.0 183 Session Progress
>> >> > Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604
>> >> > To: <sip:359883409291@<OPENSIPS_DOMAIN>:5060>;tag=B2B.113.667
>> >> > From: "359883327749"
>> >> > <sip:359883327749@<OPENSIPS_DOMAIN>:5060>;tag=1040
>> >> > Call-ID: 1296636915-3604-SALASWORK at 192.168.1.2
>> >> > CSeq: 361 INVITE
>> >> > Content-Type: application/sdp
>> >> > Contact: <sip:<OPENSIPS_IP>:5060;transport=udp>
>> >> > Server: OpenSIPS (1.6.3-notls (x86_64/linux))
>> >> > Content-Length: 184
>> >> >
>> >> > v=0
>> >> > o=SBCSIPUAS 900116523 1 IN IP4 <THIRD_PARTY_SIP_PROXY_IP>
>> >> > s=SBCSIPUAS SIP STACK v1.0
>> >> > c=IN IP4 <THIRD_PARTY_SIP_PROXY_IP>
>> >> > t=0 0
>> >> > m=audio 17900 RTP/AVP 0
>> >> > a=rtpmap:0 PCMU/8000
>> >> > a=sendrecv
>> >> > a=maxptime:30
>> >> >
>> >> >
>> >> > As can be seen, the media IP is not rewritten by the opensips and the
>> >> > IP
>> >> > passed to my soft phone is the IP of the termination IP for the
>> >> > opensips
>> >> > (i.e. the third party SIP proxy IP). Because of that, my soft phone
>> >> > starts
>> >> > the RTP directly to my provider instead trough the RTP proxy that is
>> >> > attached to the opensips.
>> >> > Just to clarify, the media IP of my soft phone is not passed to my
>> >> > provider
>> >> > - that case is handled good.
>> >> >
>> >> >
>> >> > Any idea what is missing ?
>> >> >
>> >> > Thanks in advance.
>> >> > -- Kamen
>> >> >
>> >> > _______________________________________________
>> >> > Users mailing list
>> >> > Users at lists.opensips.org
>> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >> >
>> >> >
>> >>
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>> >
>> >
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>> >
>> >
>>
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