MRTC WebRTC-SIP Gateway
Quick Start Guide
The Mizu WebRTC-SIP Gateway (MRTC) is a full stack protocol
converter between WebRTC and SIP, including all the modules needed for optimal
signaling and media conversion (ICE, TURN and STUN are built-in). Using this
software you can initiate and receive calls with WebRTC clients (usually
running in browsers) via your existing SIP server.
MRTC can be installed on any Windows
OS and it runs as a Windows service (NT service).
You should install it on a Server or
PC close to your existing SIP server (Softswitch or IP-PBX). It can be run also
from a virtual machine.
For up to 100 simultaneous calls any
PC is fine which can barely run Windows, such as a dual core Xeon with 4 GB RAM
and 30 GB free disk space. See the requirements if you have more traffic.
Follow these steps to get started:
1) Download the MRTC installer from here (this is the free version for up to
20 users and 5 simultaneous calls)
2) Double-click to start the install
process and follow the instructions (requires Administrator rights)
3) Follow the Configuration Wizard:
Once the install completes, it should
automatically start the MManage admin client with its Configuration Wizard. Otherwise
launch the “MManage.exe” application and go to “Tools” menu -> “Server setup”
-> “Configuration Wizard”:
1.
Follow the
“Quick/Auto configurations” Wizard type for the easiest setup
2.
Take care of the
“Bind IP” and “Public IP” settings if your server has multiple networks or you
are behind NAT
3.
Set a domain name
and select the “Auto SSL” checkbox if you need secure websocket (WSS). WebRTC
clients from Chrome browsers will not work if you don’t host your webpage on
HTTPS or you MRTC gateway don’t use WSS.
4.
If your
WebRTC-SIP gateway is behind NAT, auto SSL will work only if you forward ports
80 and 443 on your router from the internet (These ports are required to
acquire a “Let’s Encrypt” certificate).
5.
If you haven’t
set a domain or SSL, then you can still use Firefox to run your WebRTC client
as this browser doesn’t require HTTPS/WSS.
6.
Make sure that
the ports used by the WebRTC-SIP proxy (SIP port, Access port, Secure port) are
not used by some other application such as a local web server (in this case
either change the MRTC ports or the third party app port or bind them to
separate IP address)
7.
Set the upper
server to your existing SIP server address (also set the :port if your server
is not using the standard 5060 UDP port)
8.
Click “Next” and
“Apply” to save the settings
4) Note:
·
You don’t need to
change any settings on your existing SIP server
·
You don’t need to
manage users/extensions on the Gateway (manage them on your SIP server as you
did it before)
·
The gateway can
be also used with more than one SIP server. See the Guide if you wish to use
one gateway with multiple SIP servers.
·
There is no any
maintenance required by the MRTC gateway. Once configured properly and started,
it will run forever without the need for any maintenance work as it will
self-manage itself (including deleting old logs and auto-adapting to
environment and network conditions)
5) Start the gateway service if not already started. This
can be done from:
·
MManage ->
Control menu -> Start
·
Or from Windows
Service manager (services.msc) -> “mserver” entry (right click and select
“Start”)
Your WebRTC-SIP proxy is ready to accept connections and
calls at this stage
6) Configure your WebRTC client:
·
To find out how
to configure your WebRTC client, go to “Help” menu -> “How to connect?”.
This will display clear and easy to follow instructions about how exactly you
will have to configure your WebRTC client.
7) First quick test call:
1.
Open your browser
with your favorite WebRTC client, enter the settings from the above mentioned
settings with SIP account A (username/password valid on your SIP server) and
connect/register to your SIP server via the MRTC gateway
2.
Open your
favorite SIP client and register directly to your SIP server with
account/extension B
3.
Make calls
between A and B
Note:
·
Calls to
outbound/PSTN/carrier will be handled in the exact same way as the above
WebRTC->SIP call
·
You can also make
WebRTC to WebRTC calls (both endpoint running from browsers with SIP
credentials valid on your softswitch/IP-PBX)
8) More:
Following the above steps might
fulfill most of your needs as the gateway will auto configure and fine-tune all
its modules for optimal WebRTC-SIP protocol conversion out of the box. However
there are a lot more you can do with the gateway such as using it with multiple
SIP servers, optimize PBX features, run health analysis, setup VoIP push
notifications, export CDR records, handle special NAT requirements or optimize
(avoid) codec transcoding.
For more advanced needs, you can:
·
Re-run the configuration
wizard in “Detailed wizard” mode
·
Change any
settings from the “Configuration” form
·
Check the other
built-in modules (open various forms in MManage)
·
Check the Guide
·
WebRTC-SIP Gateway home page
·
Documentation
·
Licensing
·
Contact
Copyright © Mizutech SRL