<div dir="ltr">it's posible dispatching websocket request?<div><br></div><div>I am try to connect browser(WebRTC) to sip-phone and vice versa,How i can achieve more concurrent call(more then 1000 call)</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Jun 13, 2015 at 8:49 PM, Alex Balashov <span dir="ltr"><<a href="mailto:abalashov@evaristesys.com" target="_blank">abalashov@evaristesys.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="background-color:rgb(255,255,255);line-height:initial">                                                                                      <div style="width:100%;font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)">That question is difficult to answer without some elaboration on your part as to what you want to achieve.</div>                                                                                                                                     <div style="width:100%;font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)"><br style="display:initial"></div>                                                                                                                                     <div></div>                                                              <div style="font-size:initial;font-family:Calibri,'Slate Pro',sans-serif;color:rgb(31,73,125);text-align:initial;background-color:rgb(255,255,255)">--<br>Alex Balashov | Principal | Evariste Systems LLC<br>303 Perimeter Center North, Suite 300<br>Atlanta, GA 30346<br>United States<br><br>Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)<br>Web: <a href="http://www.evaristesys.com/" target="_blank">http://www.evaristesys.com/</a>, <a href="http://www.csrpswitch.com/" target="_blank">http://www.csrpswitch.com/</a><br><br>Sent from my BlackBerry.</div>                                                                                                                                                                                  <table width="100%" style="background-color:white;border-spacing:0px"> <tbody><tr><td colspan="2" style="font-size:initial;text-align:initial;background-color:rgb(255,255,255)">                           <div style="border-style:solid none none;border-top-color:rgb(181,196,223);border-top-width:1pt;padding:3pt 0in 0in;font-family:Tahoma,'BB Alpha Sans','Slate Pro';font-size:10pt">  <div><b>From: </b>Murugan Pandian</div><div><b>Sent: </b>Saturday, June 13, 2015 09:47</div><div><b>To: </b><a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a></div><div><b>Reply To: </b>Kamailio (SER) - Users Mailing List</div><div><b>Subject: </b>[SR-Users] SIP-over-Websocket Load Balancing</div></div></td></tr></tbody></table><span class=""><div style="border-style:solid none none;border-top-color:rgb(186,188,209);border-top-width:1pt;font-size:initial;text-align:initial;background-color:rgb(255,255,255)"></div><br><div><div dir="ltr">HI,<div><br></div><div>  how to handle sip-over-websocket load balancing (WebRTC)</div></div>
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