Probably you will get sip re invite only if you will use "<span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">bypass_media_<b>after_bridge</b>"<b> </b>variable<b>.</b></span><div>
<font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;">Vitalie</span></font></div>
<div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></font><br><div class="gmail_quote">2010/6/8 Phillip Jones <span dir="ltr"><<a href="mailto:pjintheusa@gmail.com">pjintheusa@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Ok so I set up a simple test:<br><br><action application="set" data="bypass_media=true"/><br>
<action application="bridge" data="bridge,sofia/gateway/broadvox/...."/><br><br>
and yes you appear to correct David - no SIP re-invite is involved - no RTP streams hit the box at all.<br><br>nice.<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Mon, Jun 7, 2010 at 5:31 PM, Code Ghar <span dir="ltr"><<a href="mailto:codeghar@gmail.com" target="_blank">codeghar@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">I haven't used bypass-media extensively, either. So I am also interested in knowing if it would introduce re-invite. I know several carriers, using some older SIP implementations or for other reasons, do not support re-invite. In a perfect world we would use bypass-media on the ingress and egress FS-SIP servers, without introducing re-invite, and handle media using intermediate FS-RTP servers.<br>
<br>Aside from the media, if we use, say two FS-SIP servers, then all FS-RTP servers can do load balance when doing egress. In this way we only need two signaling IPs for all kinds of customers and carriers and don't have to differentiate between ingress-only and egress-only FS-SIP servers. Each FS-SIP is ingress and egress while all FS-RTP look only like media IPs to all customers and carriers. Of course, I would like to get consensus confirmation from experienced users that bypass-media does not introduce re-invite.<br>
<br>We could use a SIP Proxy, as advised by PJ, but if a bunch of FS servers could do that same job that would be awesome, too.<div><div></div><div><br><br><br><br><div class="gmail_quote">On Mon, Jun 7, 2010 at 12:34 PM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex"><div style="word-wrap:break-word">Phillip,<div><br></div><div>please do :)</div><div><br>
</div>
<div>Well, I could be wrong, but this setup should not require any re-invite.</div><div>I never really used bypass-media on FS, but from what I understood, it will jut advertise the customer IP to FS-RTP-3 and FS-RTP-3's IP to the customer.</div>
<div><br></div><div>Anyway, the idea of the design was an attempt to answer to Code's question at the beginning of the thread. who wanted to build a such architecture with FS only.</div><div><div><br><div> <span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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<div><font face="'Helvetica Neue'"><font color="#1c00ff">David Ponzone </font><font color="#000000" size="3"><span style="font-size:12px">Direction Technique</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size:13px">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
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Le 07/06/2010 à 19:02, Phillip Jones a écrit :</div><div><div></div><div><br><blockquote type="cite">David,<br><br>I hope you don't mind me interjecting here. But what is the advantage of your setup over the traditional SIP Proxy - FS - SIP Proxy setup? Isn't introducing a re-invite here, to shift the media from FS-SIP-Internal-1 to FS-RTP-3 introducing a further complication and potential point of failure?<br>
<br>Pj<br><br><div class="gmail_quote">On Mon, Jun 7, 2010 at 7:29 AM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
<div style="word-wrap:break-word">Mike,<div><br></div><div>You're right, it can be achieved with SIP now that I think a bit more about it.</div><div>The idea was to allow adding multiple media gateways when required, so the media gateways should not be facing the carriers as some of them do SIP-filtering, but should only be advertised in the SDP.</div>
<div><br></div><div>So SIP-only boxes (doing bypass-media) should face the carriers to handle the trunking.</div><div>In the middle, we would then have the media gateways, doing SIP and mostly RTP.</div><div>But I guess we dont want customers to register and to send calls to a media gateway, so we need another set of SIP boxes on the other side, doing bypass-media also.</div>
<div><br></div><div>So it would like this:</div><div><br></div><div> ------sip-----FS-RTP-1-----sip------</div><div>FS-SIP-Internal-1 ------sip-----FS-RTP-2-----sip------FS-SIP-External-1----sip-----Carriers</div>
<div> ------sip-----FS-RTP-3-----sip------</div><div><div>FS-SIP-Internal-2 -------sip----FS-RTP-4-----sip------FS-SIP-External-2-----sip----Carriers</div><div> -------sip----FS-RTP-5-----sip------</div>
<div><br></div><div>Thanks to bypass-media, the RTP streams would go from customer to FS-RTP-x to Carriers, and reverse.</div><div>And I don't see any reason why the same set of FS-SIP boxes could not be used for both internal and external borders.</div>
<div><br></div><div>Is there something wrong in this ?</div><div><br></div><div>Code, does it help ?</div><div><br></div></div><div><div><div> <span style="border-collapse:separate;color:rgb(0, 0, 0);font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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Le 05/06/2010 à 19:54, Michael Jerris a écrit :</div><br><blockquote type="cite"><div><div></div><div><div style="word-wrap:break-word">Why would it be an advantage to have your media proxies use another protocol?<div>
<br></div><div>Mike<br><div><br><div><div>On Jun 4, 2010, at 1:59 AM, David Ponzone wrote:</div><br><blockquote type="cite"><div style="word-wrap:break-word">It doesn't solve the issue that all the media servers will do signaling too, and will talk SIP with the carriers.<div>
So the carriers will need to allow all the media servers .</div><div><br></div><div>The only clean solution to avoid that, I think, is to have signaling boxes allocating resources from media servers with another protocol than SIP.</div>
<div>RTPproxy does that I think, but I am not sure how it works.</div><div><br><div> <span style="border-collapse:separate;font-family:Helvetica;font-size:14px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><div style="word-wrap:break-word">
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