VoIP Mailing List Archives :: View topic - [asterisk-users] WebRTC phone

Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community

[asterisk-users] WebRTC phone

VoIP Mailing List Archives Forum Index -> Asterisk Users

View previous topic :: View next topic

Author

Message

cursor at telecomabmex...
Guest

Posted: Thu Feb 26, 2015 5:24 pm Post subject: [asterisk-users] WebRTC phone

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 México S.A. de C.V. Carlos Chávez +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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

jarrod at mogl.com
Guest

Posted: Thu Feb 26, 2015 5:55 pm Post subject: [asterisk-users] WebRTC phone

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@telecomabmex.com (cursor@telecomabmex.com)> wrote:

Quote:

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 México S.A. de C.V.
Carlos Chávez
+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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
********** Confidentiality Notice **********This electronic transmission and any attached documents or other writings are confidential and are for the sole use of the intended recipient(s) identified above. This message may contain information that is privileged, confidential or otherwise protected from disclosure under applicable law. If the receiver of this information is not the intended recipient, or the employee, or agent responsible for delivering the information to the intended recipient, you are hereby notified that any use, reading, dissemination, distribution, copying or storage of this information is strictly prohibited. If you have received this information in error, please notify the sender by return email and delete the electronic transmission, including all attachments from your system.

paul.belanger at polyb...
Guest

Posted: Wed Mar 04, 2015 11:08 am Post subject: [asterisk-users] WebRTC phone

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod@mogl.com> wrote:

Quote:

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.

+1 to everything here. We also do this and it works quiet well. Kudos.

--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

jarrod at mogl.com
Guest

Posted: Wed Mar 04, 2015 2:07 pm Post subject: [asterisk-users] WebRTC phone

I'll also warn that if you do intend on doing anything with rtpengine/webrtc2sip etc. you'll need Openssl 1.0.1j or better so that it has the proper support for DTLS-SRTP. This means you are sort of SOL if you are running CentOS5 unless you plan on building Openssl 1.0.1 manually. The websocket module is also not in the Kamailio repositories for CentOS5 and you would need to build it yourself. I'm being rather specific with Cent/RHEL just because I ran into this problem and wanted to throw it out as a warning...

Also rtpengine won't work in kernel mode in Cent/RHEL 5. Everything is smooth sailing on CentOS 6.

Smile

On Wed, Mar 4, 2015 at 8:07 AM, Paul Belanger <paul.belanger@polybeacon.com (paul.belanger@polybeacon.com)> wrote:

Quote:

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod@mogl.com (jarrod@mogl.com)> wrote:

Quote:

+1 to everything here. We also do this and it works quiet well. Kudos.

--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com (paul.belanger@polybeacon.com) | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

--
_____________________________________________________________________
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Display posts from previous:

	VoIP Mailing List Archives Forum Index -> Asterisk Users	All times are GMT - 5 Hours
Page 1 of 1

You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum