[OpenSIPS-Users] OpenSips as an inbound loadbalancer and outbound proxy - issue with routing BYES

[Matt Stockton]

After doing some reading, thinking that I may be able to solve my problem
using the dialog module and fix_route_dialog(). Since the Contact Header is
being re-written by the Provider, will fix_route_dialog() change it on the
way in to the appropriate value in the dialog, causing the message to be
forwarded correctly? The code for what I'm thinking is below.
Unfortunately, the issue is so intermittent that I have not been able to
re-produce the case of the Provider re-writing the Contact Header since I
started trying this modification. Does anyone know if this strategy will
work?

       if (!has_totag()) {
                create_dialog();
                # initial request
                record_route();
        }
        else {
                # sequential request - obey the indicated route
                loose_route();

                if ($DLG_status!=NULL) {
                   if (!validate_dialog()) {
                      fix_route_dialog();
                   }
                }

                t_relay();
                exit;
        }


On Fri, Nov 11, 2011 at 9:23 AM, Matt Stockton <mstockton at harqen.com> wrote:

> I added the following to my config:
>
> onreply_route[pstn_outgoing] {
>    setflag(1);
>    if(t_check_status("200")) {
>      if(search("Content-type: application/sdp")) {
>         fix_nated_sdp("8", "x.x.x.x");
>      }
>    }
> }
>
> And the only thing that seemed to have changed in the SIP signalling is,
> when my provider sends SIP 200 OK to OpenSips, OpenSips is just re-writing
> the OK Content before sending onto Freeswitch. It doesn't look like any
> messages have changed that are going from OpenSips to the provider, and the
> provider is still sending the BYE message with the Contact Header
> intermittently to the different IPs as I described. Any more thoughts on
> this?
>
> My end goal is to simply have the same OpenSips act as both an inbound
> proxy and outbound proxy, with no registration requirements. If there is an
> easier way to do this with OpenSips, please let me know. I've looked around
> and checked out all the referenced tutorials on the site w/r/t load
> balancing and routing
>
> v=0
> o=Sonus_UAC 14270 3732 IN IP4 x.x.x.x
> s=SIP Media Capabilities
>  c=IN IP4 67.231.0.110
> t=0 0
> m=audio 25620 RTP/AVP 0 101
>  a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>  a=sendrecv
> a=maxptime:20
> a=oldmediaip:ACTUAL_IP
>
>
>
> On Thu, Nov 10, 2011 at 7:17 PM, ddgiants <ddgiants at gmail.com> wrote:
>
>> Matt,
>> Sounds like you may not be topology hiding the 200OKs. Some/most SIP
>> devices/SBC's etc will send a BYE to the ip address in the o(owner) line
>> of
>> the SDP of the 200OK. Some devices do not and send BYE to the
>> initator/proxy
>> they are configured for. The 200OK will have o and c. O is the owner,
>> meaning signaling owner and C is the connection IP as in where to send the
>> actual RTP. I use the below to change the o portion of the SDP in 200 OKs
>> and works for me. Give it a try and let me know.
>>
>> onreply_route[1] {
>>  setflag(1);
>>  if(t_check_status("200")) {
>>    if(search("Content-type: application/sdp")) {
>>      fix_nated_sdp("8", "x.x.x.x");
>>    }
>>  }
>> }
>>
>> DD
>>
>> --
>> View this message in context:
>> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSips-as-an-inbound-loadbalancer-and-outbound-proxy-issue-with-routing-BYES-tp6984017p6984067.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
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