HELLO & WELCOME

[WelshPaul]

What is Call Encryption, TLS and SRTP?
Call Encryption is a method of encrypting both your VoIP SIP traffic (The handshake that introduces and closes a call) and your actual VoIP Audio, often referred to as RTP traffic.

Yay.com support turning on both TLS (Transport Layer Security) to encrypt your VoIP SIP traffic and turning on encryption for your RTP traffic to make the actual audio secure using SRTP (Secure RTP). To activate TLS for the SIP traffic you don't need to do anything in your Yay.com dashboard.

SRTP encryption deals with encrypting the actual audio of the call. Therefore it can be turned on for both the inbound call audio and the outbound call audio. So for listening to what your caller has to say (inbound) and you talking to the caller (outbound). Below is what you need to do to encrypt each part:

Configure your FreePBX trunk as follows...

Outgoing

Trunk Name:

Code: Select all

Yay-your_SIP_username

PEER Details:

Code: Select all

type=peer
qualify=yes
transport=tls
username=your_SIP_username
secret= your_SIP_password
host=talk.yay.com
dtmfmode=rfc2833
disallow=all
allow=alaw
encryption=yes

Incoming

USER Context:

Code: Select all

from-trunk-Yay-your_SIP_username

USER Details:

Code: Select all

type=peer
qualify=yes
nat=yes
insecure=port,invite
host=talk.yay.com
disallow=all
allow=alaw
transport=tls
encryption=yes

Register String:

Code: Select all

tls://your_SIP_username:[email protected]/your_SIP_username

**Simply replace "your_SIP_username" with your SIP username and "your_SIP_password" with your SIP password.

Once you've done that, you then need to enable call encryption in your Yay.com Dashboard. To do this go to your Dashboard > MyVoIP > User > Sip User > Edit > Call Encryption > Enable.

Example: