Unfortunately FS does not (as yet) support it. At least not AFAIK.<div>It also would require that both endpoints support it too, and not all do.<div><br></div><div>In any case since FS is a B2BUA not a SIP proxy the endpoints are negotiating with FS not the other endpoint - so you're still going to want to detect that you want to (try to) bypass_media.</div>
<div><br></div><div>-Steve</div><div><br></div><div><br><div><div><br></div><div><br><br><div class="gmail_quote">On 11 January 2013 11:19, Raimund Sacherer <span dir="ltr"><<a href="mailto:raimund.sacherer@logitravel.com" target="_blank">raimund.sacherer@logitravel.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:10pt;font-family:arial,helvetica,sans-serif"><div style="font-size:10pt;font-family:arial,helvetica,sans-serif">
Hi, <br><br>isn't there allready a protocol that does that? ICE?<br><br>this is an excerpt from the SNOM wiki page (we have about 280 snom phones in action):<br><br>--- snip ---<br>ICE DESCRIPTION
<dd>
</dd><dd> Choose whether or not you want to use ICE (Interactive
Connectivity Establishment). ICE optimizes the media path. This would be
the case, for example, when two phones in the same network are calling
each other via a long media path through other, external networks. With
ICE, the short media path in the same network would be chosen, which
will presumably have better quality than the long one. Sometimes this
feature will stop you from being able to make calls. When this occurs,
switch it off.
</dd><br>--- snip ---<br><br>Link here: <a href="http://wiki.snom.com/Settings/user_ice" target="_blank">http://wiki.snom.com/Settings/user_ice</a><br><br>Here seems to be more information, check "Probing Media Path": <a href="http://wiki.snom.com/Networking/NAT" target="_blank">http://wiki.snom.com/Networking/NAT</a><br>
<br>Best, <br><br>Raimund<br><br><hr><div style="font-size:12pt;font-style:normal;font-family:Helvetica,Arial,sans-serif;text-decoration:none;font-weight:normal"><b>From: </b>"Chris B. Ware" <<a href="mailto:chrisbware@yahoo.it" target="_blank">chrisbware@yahoo.it</a>><br>
<b>To: </b><a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br><b>Sent: </b>Viernes, 11 de Enero 2013 10:30:52<br><b>Subject: </b>[Freeswitch-users] RTP traffic on LAn<div class="im">
<br><br><div style="font-size:12pt;font-family:times new roman,new york,times,serif"><div>Hi all,</div><div><br></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
I'm trying to find a way to let RTP traffic between two phones, registered to a public Freeswitch, on the same LAN, remain local.</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
Usually phones are natted behind an ADSL router and using two RTP streams to speack each other (because, for example, are on different </div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
rooms on the same office) consume a lot of
bandwidth.</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif"><br></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
Is there anyone who has found a solution to this problem?</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif"><br></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
I was thinking on using SIP INFO for sending DTMF (in case customer have to use some phone features activated via phone), using</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
private IP (no STUN on phones) and keeping FS out of RTP streaming.</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif"><br>
</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">Any help will be appreciated.</div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
<br></div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">Chris </div><div style="font-style:normal;font-size:16px;background-color:transparent;font-family:'times new roman','new york',times,serif">
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