[OpenSIPS-Users] OpenSIPs behind a firewall

[Daniel Goepp]

Sorry, I hit send when I meant to hit save while I pulled some info.

The trace below though shows the start of the loop of ACKs.  The ACKs are
getting sent to 75.101.136.125, when then should be going to the endpoint.
It would appear the transaction is not being identified.  I'm still looking
at why, hence not wanting to send this yet...but any early ideas would be
great ;)

-dg


On Wed, Oct 14, 2009 at 4:20 PM, Daniel Goepp <dan at goepp.net> wrote:

> I'm sure that Raul had a good point that I'm crazy for doing this, but
> right now unfortunately I do not have a lot of options, and believe with
> some solid logic in the config can resolve this issue.  Right now I appear
> to be having an ACK come back from a 200 that turns into an endless loop.
> I'm using
>
> advertised_address="75.101.136.125"
>
>
>
> U 10.250.7.164:5060 -> 76.102.118.209:3724
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 192.168.1.102:3724
> ;received=76.102.118.209;branch=z9hG4bK-d8754z-ff34ad60b13c376a-1---d8754z-;rport=3724.
> Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E..
> CSeq: 2 INVITE.
> Contact: <sip:2001 at 76.102.118.209:11386>.
> From: "Daniel Goepp" <sip:1001 at vidtel.com <sip%3A1001 at vidtel.com>
> >;tag=ac7fa632.
> To: "2001" <sip:2001 at vidtel.com <sip%3A2001 at vidtel.com>
> >;tag=8f7b7e78f8cb0eca.
> Record-Route: <sip:75.101.136.125;lr=on>.
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY.
> Server: TANDBERG/257 (TE2.0.0.191892).
> Supported:
> replaces,100rel,timer,gruu,path,outbound,com.tandberg.sdp.extensions.v1.
> Session-Expires: 500; refresher=uas.
> Min-SE: 90.
> Content-Type: application/sdp.
> Content-Length: 217.
> .
> v=0.
> o=tandberg 103 1 IN IP4 192.168.1.101.
> s=-.
> c=IN IP4 192.168.1.101.
> b=CT:64.
> t=0 0.
> m=audio 2330 RTP/AVP 0 101.
> b=TIAS:64000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> U 76.102.118.209:3724 -> 10.250.7.164:5060
> ACK sip:2001 at 76.102.118.209:11386 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.102:3724
> ;branch=z9hG4bK-d8754z-6d520059c54f5b74-1---d8754z-;rport.
> Max-Forwards: 70.
> Route: <sip:75.101.136.125;lr>.
> Contact: <sip:1001 at 192.168.1.102:3724>.
> To: "2001"<sip:2001 at vidtel.com <sip%3A2001 at vidtel.com>
> >;tag=8f7b7e78f8cb0eca.
> From: "Daniel Goepp"<sip:1001 at vidtel.com <sip%3A1001 at vidtel.com>
> >;tag=ac7fa632.
> Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E..
> CSeq: 2 ACK.
> Proxy-Authorization: Digest username="1001",realm="vidtel.com
> ",nonce="4ad65ba02105a03bfdc0e3839160a42b5e1d90ac",uri="
> sip:2001 at vidtel.com <sip%3A2001 at vidtel.com>
> ",response="0eea648b121320214ab8ec908eb97446",algorithm=MD5.
> User-Agent: eyeBeam release 1104g stamp 54685.
> Content-Length: 0.
> .
>
>
> U 10.250.7.164:5060 -> 75.101.136.125:5060
> ACK sip:2001 at 76.102.118.209:11386 SIP/2.0.
> Via: SIP/2.0/UDP 75.101.136.125;branch=z9hG4bK3128.7db9df05.2.
> Via: SIP/2.0/UDP 192.168.1.102:3724
> ;received=76.102.118.209;branch=z9hG4bK-d8754z-6d520059c54f5b74-1---d8754z-;rport=3724.
> Max-Forwards: 69.
> Route: <sip:75.101.136.125;lr>.
> Contact: <sip:1001 at 192.168.1.102:3724>.
> To: "2001"<sip:2001 at vidtel.com <sip%3A2001 at vidtel.com>
> >;tag=8f7b7e78f8cb0eca.
> From: "Daniel Goepp"<sip:1001 at vidtel.com <sip%3A1001 at vidtel.com>
> >;tag=ac7fa632.
> Call-ID: NTU3NTE3NmZiNjVkNGYzNWEzYWVhZWQ4MjhhYjczN2E..
> CSeq: 2 ACK.
> Proxy-Authorization: Digest username="1001",realm="vidtel.com
> ",nonce="4ad65ba02105a03bfdc0e3839160a42b5e1d90ac",uri="
> sip:2001 at vidtel.com <sip%3A2001 at vidtel.com>
> ",response="0eea648b121320214ab8ec908eb97446",algorithm=MD5.
> User-Agent: eyeBeam release 1104g stamp 54685.
> Content-Length: 0.
> .
>
> -dg
>
>
>
> On Wed, Oct 7, 2009 at 11:20 PM, Bogdan-Andrei Iancu <
> bogdan at voice-system.ro> wrote:
>
>> Hi Daniel,
>>
>> Try using set_advertised_address() before sending the call out.
>>      http://www.opensips.org/Resources/DocsCoreFcn#toc125
>>
>> or use the record_route_preset() function :
>>     http://www.opensips.org/html/docs/modules/devel/rr.html#id228590
>>
>> Regards,
>> Bogdan
>>
>> Daniel Goepp wrote:
>> > After further investigation, this only updating the Via header, but
>> > the RR is remaining untouched:
>> >
>> > Record-Route: <sip:10.250.7.164;lr=on>
>> >
>> > Ideas about how I might get this field updated correctly?
>> >
>> > Thanks
>> >
>> > -dg
>> >
>> > On Wed, Oct 7, 2009 at 2:10 PM, Bogdan-Andrei Iancu
>> > <bogdan at voice-system.ro> wrote:
>> >
>> >> Hi Daniel,
>> >>
>> >> try using the advertise_address and advertise_port to force opensips in
>> >> using the public IPs of the NAT:
>> >>    http://www.opensips.org/Resources/DocsCoreFcn#toc24
>> >>
>> >> Regards,
>> >> Bogdan
>> >>
>> >> Daniel Goepp wrote:
>> >>
>> >>> I am trying to setup OpenSIPs to run behind a firewall, and not
>> >>> finding much information regarding how to get OpenSIPs to be aware of
>> >>> the public IP for it's signaling.  The firewall is setup with a 1 to 1
>> >>> NAT for the public and private IPs, and right now all udp and tcp
>> >>> traffic is being passed directly through.  Has anyone successfully
>> >>> gotten OpenSIPs setup like this?  If so, can you please provide any
>> >>> information on how it was setup.
>> >>>
>> >>> TIA
>> >>>
>> >>> -dg
>> >>>
>> >>> _______________________________________________
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>> >>> 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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>
>
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