<div dir="ltr">Great doc!<div><br></div><div>Some short notes:<br><div><br></div><blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<b id="internal-source-marker_0.27374149672687054" style="font-family:'Times New Roman';font-size:medium"><span style="font-size:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-space:pre-wrap">This is broken. Cisco is the most well known vendor to make this mistake. These implementations should be fixed to use G729 as the codec name with payload type 18.</span></b></blockquote>
<div>Linksys / Sipura SPAs do this, too, I'm pretty sure even before they were bough by cisco.</div><div><br></div><blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<b id="internal-source-marker_0.27374149672687054" style="font-family:'Times New Roman';font-size:medium"><span style="font-size:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-space:pre-wrap">Typically this NAT functionality is referred to as a SIP NAT helper, or SIP ALG. Of course there are other names.</span></b></blockquote>
<div>Many routers let you turn off NAT ALG, if you can find it in the menu.</div><div><br></div><blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<b id="internal-source-marker_0.27374149672687054" style="font-family:'Times New Roman';font-size:medium"><span style="font-size:15px;font-family:Arial;font-weight:normal;vertical-align:baseline;white-space:pre-wrap">Latching has several issues:... scalability</span></b></blockquote>
<div>At what point does this seriously become an issue? 100 concurrent calls? 1000? If you're proxying media anyway the IP rewriting sounds pretty minimal.</div><div><br></div><div>More info about NAT, one way media, and how sipsorcery is a SIP proxy that doesn't ever proxy media but can still set up calls with 2 way media would be much appreciated.</div>
<div><br></div><div>One other things I'm interested in is the ability to not handle media, for latency & performance reasons:</div><div>What are the pitfalls with FreeSWITCH specifically that makes it not happy with not handling media?</div>
<div>and.. if I still need FS to handle DTMF what are my options? Do many/most carriers (on origination) support SIP INFO or something like that? (I run a calling card with an option to hang up the call)</div><div><br></div>
<div><br></div><div>Thanks for the helpful info!</div><div><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi </span></span></div>
<br><br><div class="gmail_quote">On Fri, Jun 15, 2012 at 8:24 PM, dingdong <span dir="ltr"><<a href="mailto:freeswitch@zoho.com" target="_blank">freeswitch@zoho.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u><div><div style="font-size:10pt;font-family:Verdana,Arial,Helvetica,sans-serif">very nice and helpful,and yes,it would really be more useful if somehow related to how does FS do SIP.<div><br></div><div>Thanks<br><div>
<br>---- On Thu, 14 Jun 2012 13:06:52 -0700 <b>Kristian Kielhofner <<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></b> wrote ---- <br></div><div><div class="h5"><br><blockquote style="border-left:1px solid #0000ff;padding-left:6px">
FreeSWITCHers, <br> <br> As some of you may have noticed I frequently comment on SIP-related <br>FreeSWITCH posts. Long story short I decided to finally put some of <br>these thoughts and experiences down in writing. What I ended up with <br>
is a 21 page (and counting) document describing various SIP header <br>fields, SDP information, RTP issues, DTMF issues, NAT traversal <br>technologies, interop headaches, etc. <br> <br> I wrote a little intro and linked to the document if any of you <br>
would like to check it out (link at the end of post): <br> <br><a href="http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html" target="_blank">http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html</a> <br>
<br> Of course I'd appreciate any feedback you may have. What do you <br>disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec <br>questions you'd like to see me address? Should I get into more <br>
implementation (FreeSWITCH) specific issues? Let me know. <br> <br>Thanks! <br> <br>-- <br>Kristian Kielhofner <br> <br>_________________________________________________________________________ <br>Professional FreeSWITCH Consulting Services: <br>
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<br></blockquote></div><br></div></div></div>