<div dir="ltr">For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages:<div><br></div><div><div>kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-carrierroute.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-cpl.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-debuginfo.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-gzcompress.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-ims.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-ldap.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-mysql.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-outbound.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-postgres.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-presence.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-python.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-sctp.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-snmpstats.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-sqlite.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-tls.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-unixodbc.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-utils.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-uuid.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-websocket.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-xml.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div><div>kamailio-xmpp.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms</div></div><div><br></div><div>Keep in mind that using Kamailio to bridge the signalling is only half of the equation. You must also bridge the media and so the rtpengine module allows Kamailio to interface with the rtpengine (<a href="https://github.com/sipwise/rtpengine">https://github.com/sipwise/rtpengine</a>) which does that half.</div><div><br></div><div>In the provided example Kamailio.cfg there isn't any real hardening and it's pretty much purely used as a bridge that would front an Asterisk 1.8 server for webrtc capabilities (but not any other sip). It uses the dispatcher module to dispatch to the underlying asterisk so you will still need to add the Asterisk to the dispatcher config.</div><div><br></div><div>Thanks,</div><div>Jarrod</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 26, 2015 at 2:55 PM, Jarrod Cuzens <span dir="ltr"><<a href="mailto:jarrod@mogl.com" target="_blank">jarrod@mogl.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><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"><div><div class="h5"><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><font color="#888888"><br>
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