<div dir="ltr"><div>For the client:</div>JSSIP and Sipml5.<div><br></div><div>If you are going to be coding something up yourself I like the JSSIP 0.5.x javascript interfaces. If you are simply going to use a pre-canned one then sipml5 works pretty well and remembers your settings in localstorage. I haven't used any closed source versions since the above works really well for us.</div><div><br></div><div>For the server:</div><div>If you are using Asterisk 1.8 you'll need to front it with Kamailio and rtpengine (or webrtc2sip but I have had stability issues with that). If you are using a more recent asterisk then the webrtc is built in but I haven't used (we use Kamailio and rtpengine to bridge webrtc).</div><div><br></div><div>If you need the kamailio config I can send it to you (it gets complicated). The rtpengine works very well if you can run the kernel module and introduced very little cpu overhead.</div><div><br></div><div>Thanks,<br>Jarrod</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 26, 2015 at 2:24 PM, Carlos Chavez <span dir="ltr"><<a href="mailto:cursor@telecomabmex.com" target="_blank">cursor@telecomabmex.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"> Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source. Customers are starting to ask for web solutions and we need to start testing.<span class="HOEnZb"><font color="#888888"><br>
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