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:
kamailio.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-auth-ephemeral.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-bdb.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-carrierroute.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-cpl.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-debuginfo.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-gzcompress.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-ims.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-ldap.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-mysql.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-outbound.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-postgres.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-presence.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-python.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-sctp.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-snmpstats.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-sqlite.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-tls.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-unixodbc.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-utils.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-uuid.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-websocket.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-xml.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
kamailio-xmpp.x86_64 4.2.1-4.1
@home_kamailio_v4.2.x-rpms
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 (
https://github.com/sipwise/rtpengine) which does that half.
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.
Thanks,
Jarrod
On Thu, Feb 26, 2015 at 2:55 PM, Jarrod Cuzens <jarrod at mogl.com> wrote:
> For the client:
> JSSIP and Sipml5.
>
> 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.
>
> For the server:
> 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).
>
> 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.
>
> Thanks,
> Jarrod
>
> On Thu, Feb 26, 2015 at 2:24 PM, Carlos Chavez <cursor at
telecomabmex.com>
> wrote:
>
>> 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.
>>
>> --
>> Telecomunicaciones Abiertas de M?xico S.A. de C.V.
>> Carlos Ch?vez
>> +52 (55)9116-91161
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
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>
>
>
>
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