<div>It can be done with or without media. FS does not relay SIP messages transparently as OpenSER, but act as a SIP UA to each call leg. This gives an oportunity to place rtp in between streams, do call recording and all other softswitch stuff. I use FreeSWITCH as hosted PBX solution, basically hosting multiple SIP domains each of them representing a separate PBX for a different company with separate dialplans etc.</div>
<div> </div>
<div>Ognjen<br><br></div>
<div class="gmail_quote">On Fri, May 22, 2009 at 2:37 PM, Peter P GMX <span dir="ltr"><<a href="mailto:Prometheus001@gmx.net">Prometheus001@gmx.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">This is also interesting for me, as I love freeswitch, and maintaining a<br>single platform is easier, than handling various different ones.<br>
In the past years I did a couple of projects with OpenSER /openSIPS.<br>These projects comprised:<br><br> * registrar for the SIP user agents<br> * handle invite messages (+ ringing, bye, ok, etc also of course)<br> between registered user agents and user agents at external domains<br>
* rtp payload was a bit different from usual VoIP traffic (video<br> parts, application sharing, file downloads etc.), but SDP was fine<br> according to RFC, and OpenSER mediaproxy worked also<br> * handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS)<br>
* The number of messages to handle was not that much (some thousand<br> subs).<br><br>For my understanding this should also be possible with Freeswitch with<br>bypass_media. Right?<br><br>Best regards<br>Peter<br><br>
<br><br>Ognjen Seslija schrieb:<br>
<div class="im">> Hello,<br>><br>> FS by design is B2BUA, and it cannot route INVITEs and other SIP<br>> methods. It can however, bridge a-leg to b-leg with or w/o media and<br>> doing plenty other cool stuff much better than commercial projects. I<br>
> suggest joining us on irc to detail your setup so we can help you.<br>><br>> Regards,<br>> Ognjen (sekil on #freeswitch).<br>><br>> On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar)<br></div>
<div class="im">> <<a href="mailto:sridhart@alcatel-lucent.com">sridhart@alcatel-lucent.com</a> <mailto:<a href="mailto:sridhart@alcatel-lucent.com">sridhart@alcatel-lucent.com</a>>> wrote:<br>><br>> Hi all,<br>
><br>> I want to use FreeSWITCH as a SIP transparent proxy in session<br>> border controller application. Please let me know the changes in<br>> configuration files required to achieve this behaviour<br>
><br>> Thanks very much for the help.<br>><br>> Regards,<br>> Sridhar<br>><br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
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