SOLUTION GUIDE Wave 3 | April 2016 | 2998-0068-002 Rev B

PortSIP® WebRTC Gateway User Guide

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Visit the PortSIP® WebRTC Gateway for information on PortSIP software versions and administrative documentation, and
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Contents

®
PortSIP WebRTC Gateway User Guide .............................................................................................1
1. The WebRTC Gateway Architecture ................................................................... 4
2. How it works? ....................................................................................................... 4
3. Install WebRTC Gateway ..................................................................................... 5
4. Manage WebRTC Gateway .................................................................................. 5
4.1 Configuring WebRTC Gateway by Wizard ................................................................................. 5
4.2 Settings ......................................................................................................................................... 6
5. Run WebRTC Web Client ..................................................................................... 8
6. Deployment Practices ........................................................................................ 10
6.1 Deploy WebRTC Gateway with official certificate ................................................................... 10
6.2 Deploy WebRTC Gateway with Self-signed certificate in LAN ............................................... 11

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1. The WebRTC Gateway Architecture
Typically, the WebRTC Gateway architecture as below:

This document assumes that the Windows OS are already deployed and that Microsoft administrators
are available to administrators PortSIP WebRTC Gateway.

2. How it works?
PortSIP WebRTC Gateway sits at the network edge to bridge the traditional operator network
(PSTN/VoIP Provider/SIP Trunking/IP PBX) with the Web Browser, letting carriers build Web services
on top of it. The gateway essentially turns any Web page into a telephone that the network can dialogue
with as it would a phone.

Since the traditional operator network (PSTN/VoIP Provider/SIP Trunking) and SIP devices (IP Phone,
Softphone) don’t support the newest WebRTC standards such as TURN, DTLS-RTP, RTC-FB, ICE,
therefor we need the WebRTC Gateway to transform the signaling and RTP streams. That’s why the
Gateway is works like a bridge.

When the user make calls in Web Browser, the WebRTC Gateway will transform the Signaling to
standard SIP message and route calls to PSTN/VoIP Provider/SIP Trunking/IP PBX. After the call
established, the WebRTC Gateway receive RTP stream from browser and transcode it then send to
PSTN/VoIP Provider/SIP Trunking/IP PBX and vice versa.

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3. Install WebRTC Gateway
After you downloaded PortSIP WebRTC MSI installer just double click it and follow the instructions to
install.

You can set the WebRTC Gateway HTTP and HTTPS listen port during installation, the default HTTP
port is 9288, HTTPS ports is 9287 – Please ensure the port that doesn’t using by other
applications otherwise the HTTP/HTTPS service will fails.

4. Manage WebRTC Gateway

After the WebRTC Gateway successfully installed, you can double click the Management Console on
your Windows desktop, which will open the Management Console.

You can also use this URL remotely to manage the Gateway: http://gatweayIP:port/manager/index.html
(the gawayIP is the IP of the server which you installed the Gateway, the port is the HTTP port which
you specify during your install the gateway).

Please use below credentials to sign in the Admin Management Console:

Username: admin
Password: admin

4.1 Configuring WebRTC Gateway by Wizard

After signed in the Management Console, click the “Wizard” menu in left.

4.1.1 WebRTC settings

This page settings are for the WebRTC Gateway, please follow below instructions:

1. Click the Next button.

2. Gateway IP Address: you can specify the WebRTC Gateway IP address here, if the gateway is
run in Lan please enter local private IP address; If run on the internet please enter the static
public IP. This filed IP can not be empty.
3. Default WS port for WebRTC Gateway has been listening, and the WS transport port is 10080.
4. Listen WSS: The Google Chrome requires the WSS in order to access the camera and
microphone. You must select this option if you want use Google Chrome.

Once the “Listen WSS” is selected, you should set below fields:

WSS port: set a port which is for the WSS, for example 10443.
Domain Name: Your WebRTC Gateway domain name, it’s must resolvable, if you don’t have
domain name of Gateway server, enter the Gateway IP.
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 Certificate File: for WSS, you must upload the certificate file. You can generate the SSL
certificate file by yourself or purchase a certificate from provider such as Thawte or Digicert.
The certificate must matched the WebRTC Gateway domain.

Private Key File: It will be generated with certificate together.

Password of Private key: the password of private key file which is you entered when you
generate the certificate file. If no password for private key then leave it blank.

Note: If you haven’t a valid certificate, and don’t know how to create a Self-
signed certificate, you can upload the sample certificate(server.crt and
server.key) which is in Gateway installed folder, e.g.: “C:\Program
Files\PortSIP\WebRTC”.

Click the Next button.

4.1.2 SIP Server/PBX settings

This page settings are for the SIP Server/PBX.

1. SIP Domain: specify your SIP Domain here here, it’s can be an IP address or domain.
2. Use DNS SRV records: If your SIP Domain support DNS SRV, you can select this option.
If the DNS SRV is enabled, next steps will be skipped.
3. The server address is same as SIP domain: this option default is selected. Please un-selected
it if your SIP server address is not same as SIP Domain.
4. SIP Server Address: if your SIP Server address is not same as SIP Domain, you MUST specify
your SIP server address here.
5. SIP Transport: your SIP server transport, usually is UDP.
6. Server Port: your SIP server port.

Click the Save button to complete the wizard.

Note: after you clicked the Save button, the WebRTC Gateway will restart, you have to sign in the
management console again.

4.2 Settings
4.2.1 Log
You can enable/disable WebRTC log by select/un-select “Enable SipMessage Logging” option, and
choose the log level. The log file will be generated in WebRTC Gateway installed folder, the name is
“webrtcgw.log”. You must restart WebRTC Gateway to take effect.

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4.2.2 STUN Server
If you deploy the WebRTC Gateway in LAN and can’t access the default STUN server, you can try
install a STUN Server. Download the STUN Server then extract stunserver.zip and run stunserver.exe.

Click the “Settings” in the WebRTC Gateway Management Console and set the stun server IP as the
Run stunerver PC IP, the port default is 3478.

4.2.3 Web Server

The Web Server is use for manage WebRTC Gateway and deploy the Web Client.

https port default is: 9297

http port default is: 9298

You can change to any other port as you want. Https is request 4.1.1 step to upload a valid certificate.

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5. Run WebRTC Web Client
Assume we have a SIP Server/PBX which SIP domain is portsip.net, it has SIP user 100.

Assume we installed the WebRTC Gateway on a server which IP is 192.168.0.28.

The WebRTC Gateway setting is:

HTTP port: 9288

HTTPS port: 9287
WS port: 10080
WSS port: 10443

5.1 For Firefox:

Below settings are for last version Firefox.

1. Open http://192.168.0.28:9288/portgo/index.html by last version Firefox.

2. SIP Username filed: enter a name as 100.
3. SIP password filed: enter the password of SIP user 100.
4. SIP Domain in case is: portsip.net.
5. WS URI: in case is ws://192.168.0.28:10080 (It’s WebRTC Gateway IP and WS port).
6. Press “enter” button
7. Once the web client registered, you can make & receive calls with other SIP client.

5.2 For Firefox with HTTPS + WSS:

The Firefox also support HTTPS and WSS. In case you MUST set the HTTPS and WSS with certificate
by WebRTC Gateway Management Console Wizard.

1. Open https://192.168.0.28:9287/portgo/index.html by last version Firefox,

If you use a self-signed certificate, the Firefox alert “Your connection is not secure”, click
“Advanced”->”Add exception…”->“Confirm Security Exception”.
WSS port also need add to exception, Open https://192.168.0.28:10443 by Firefox, the Firefox
alert “Your connection is not secure”, click“Advanced”->”Add exception…”->“Confirm Security
Exception”.
2. SIP Username filed: enter a name as 100.
3. SIP password filed: enter the password of SIP user 100.
4. SIP Domain in case is: portsip.net.
5. WS URI: in case is wss://192.168.0.28:10443 (It’s WebRTC Gateway IP and WSS port).
6. Press “enter” button

Once the web client registered, you can make & receive calls with other SIP client.

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5.3 For Google Chrome:
The Google Chrome requires the HTTPS and WSS in order to access the Camera and Microphone.
You MUST set the HTTPS and WSS with certificate by WebRTC Gateway Management Console
Wizard.

1. Open https://192.168.0.28:9287/portgo/index.html by last version Google Chrome,

If you use a self-signed certificate, the Chrome alert the certificate is unsafe, just allow this
unsafe certificate (since it’s self-signed certificate)
2. SIP Username filed: enter a name as 100.
3. SIP password filed: enter the password of SIP user 100.
4. SIP Domain in case is: portsip.net.
5. WS URI: in case is wss://192.168.0.28:10443 (It is WebRTC Gateway IP and WSS port).
6. Press “enter” button

Once the web client registered, you can make & receive calls with other SIP client.

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6. Deployment Practices
6.1 Deploy WebRTC Gateway with official certificate

You can purchase an official certificate from the certificate provider such Thawte or Digicert in order to
avoid the security alert and don’t need add the security exception manually.

During you purchase the certificate, you should generate the private key and CSR by yourself – please
read the provider’s instructions or ask the provider support. We recommend you don’t set the password
for private key file. You must keep the private key file by yourself.

Assume we purchased a certificate from Thawte (in case we use the Thawte SSL123 certificate as
example) for domain examplertc.com. After you downloaded the certificate .zip file, just extract it, there
has IntermediateCA.crt and ssl_certificate.crt, you will need to use a plain text editor (for example
Windows Notepad, don’t use MS Word) to combine your two certs with them into one file:
copy all text from IntermediateCA.crt and append to ssl_certificate.crt, the ssl_certificate.crt at the top
and IntermediateCA.crt at the bottom.

1. Ensure the domain examplertc.com has been resolve correct to the server which you installed
the PortSIP WebRTC Gateway.
2. Assume the SIP server address is 69.164.210.98, the SIP Domain is: sip.testsip.com, the
transport is UDP on port 5060, it has SIP user 100, 101, 102, 103, 104.

3. Use browser open http://examplertc.com:9288/manager/index.html, sign in the Management

Console with admin/admin.
4. Enter ip of examplertc.com to “Gateway IP Address”.
5. Click “Listen WSS” to select it.
6. Set WSS port, for example: 10443
7. Set examplertc.com for “Gateway Domain Name” filed.
8. Click the up arrow icon to choose the combined “ssl_certificate.crt” file for “Certificate File”.
9. Click the up arrow icon to choose the private key file which you generated when you purchase
the certificate.
10. If you have set the password when you generate the private key, then please enter it for
“Private key password”. If don’t set the password just leave it empty.
11. Click the “Next” button.
12. Set the “SIP Domian” as sip.testsip.com
13. Un-select the “The server address is same as SIP domain”.
14. Enter the SIP Server Address, in case it is 69.164.210.98.
15. Select the SIP Transport as UDP.
16. The Server port is: 5060.
17. Click the “Save” button, the WebRTC Gateway will be restart automatically, you will need to
sign in management console again.
18. Open https://examplertc.com:9287/portgo/index.html by last version FiregFox or Google
Chrome.
19. SIP Username filed: enter a name as 100.
20. SIP password filed: enter the password of SIP user 100.
21. SIP Domain in case is: sip.testsip.com.
22. WS URI: in case is wss://examplertc.com:10443 (It is WebRTC Gateway IP and WSS port).
23. Press your “enter” button
24. Once the web client registered, you can make & receive calls with other SIP client.

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6.2 Deploy WebRTC Gateway with Self-signed certificate in
LAN

For Firefox:
Below settings are for last version Firefox.

1. Assume we have a SIP Server/PBX which IP 192.168.0.98 and SIP domain is 192.168.0.98,
this server/PBX use the UDP transport on port 5060, it has SIP user 100, 101, 102, 103, 104.
2. Assume we installed the WebRTC Gateway on a server which IP is 192.168.0.28.
3. Use browser open http://192.168.0.28:9288/manager/index.html, sign in the Management
Console with admin/admin.
4. Click the “Wizard” on left and click the “Next” button.
5. Enter 192.168.0.28 for the “Gateway IP Address” filed.
6. Click the “Next” button.
7. Enter the SIP Domain, in case it is 192.168.0.98.
8. Select the SIP Transport as UDP.
9. The Server port is: 5060.
10. Click the “Save” button, the WebRTC Gateway will be restart automatically, you will need to
sign in management console again.

11. If in LAN the WebRTC Gateway can’t access the internet, Download the STUN Server then
extract stunserver.zip and run stunserver.exe.
12. Click the “Settings” in the WebRTC Gateway Management Console and set the stun server as
192.168.0.28:3478.
13. Open http://192.168.0.28:9288/portgo/index.html by last version Firefox.
14. SIP Username filed: enter a name as 100.
15. SIP password filed: enter the password of SIP user 100.
16. SIP Domain in case is: sip.testsip.com.
17. WS URI: in case is ws://192.168.0.28:10080 (the IP is WebRTC Gateway IP and WS port).
18. Press “enter” button
19. Once the web client registered, you can make & receive calls with other SIP client.

For Google Chrome:

The Google Chrome requires the HTTPS and WSS in order to access the Camera and Microphone. We
MUST set the HTTPS and WSS with certificate for WebRTC Gateway and Gateway Web Service.

1. Use browser open http://192.168.0.28:9288/manager/index.html, sign in the Management

Console with admin/admin.
2. Enter 192.168.0.28 for the “Gateway IP Address” filed.
3. Click “Listen WSS” to select it.
4. Set WSS port, for example: 10443
5. Set 192.168.0.28 for “Gateway Domain Name” filed.
6. Click the up arrow icon to choose the “server.crt” file from the Gateway install folder, for
example: C:\Program Files\PortSIP\WebRTC.
7. Click the up arrow icon to choose the “server.key” file from the Gateway install folder, for
example: C:\Program Files\PortSIP\WebRTC.
8. Leave the “Private key password” empty.
9. Click the “Next” button.
10. Enter the SIP Domain, in case it is 192.168.0.98.
11. Select the SIP Transport as UDP.
12. The Server port is: 5060.
13. Click the “Save” button, the WebRTC Gateway will be restart automatically, you will need to
sign in management console again.
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 14. If in LAN the WebRTC Gateway can’t access the internet, Download the STUN Server then
extract stunserver.zip and run stunserver.exe.
15. Open https://192.168.0.28:9287/portgo/index.html by last version Google Chrome, when the
Chrome alert the certificate is unsafe, just allow this unsafe certificate (since it’s self-signed
certificate)
16. SIP Username filed: enter a name as 100.
17. SIP password filed: enter the password of SIP user 100.
18. SIP Domain in case is: sip.testsip.com.
19. WS URI: in case is wss://192.168.0.28:10443 (It is WebRTC Gateway IP and WSS port).
20. Press “enter” button
21. Once the web client registered, you can make & receive calls with other SIP client.

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