Secure Real-Time Transport Protocol - Yealink SIP-T48G Administrator's Manual

Administrator's Guide for SIP-T2_Series_T4_Series IP Phones
To configure the server certificates via web user interface:
1. Click on Security->Server Certificates.
2. Select the desired value from the pull-down list of Device Certificates.
3. Click Confirm to accept the change.
To upload a server certificate via web user interface:
1. Click on Security->Server Certificates.
2. Click Browse to select the certificate (*.pem and *.cer) from your local system.
3. Click Upload to upload the certificate.
A dialog box pops up to prompt "Success: The Server Certificate has been loaded!
Rebooting, please wait...".
Secure Real-Time Transport Protocol (SRTP) encrypts the RTP streams during VoIP phone
calls to avoid interception and eavesdropping. The parties participating in the call must
enable SRTP feature simultaneously. When this feature is enabled on both phones, the
type of encryption to utilize for the session is negotiated between the IP phones. This
negotiation process is compliant with RFC 4568.
When a user places a call on the enabled SRTP phone, the IP phone sends an INVITE
message with the RTP encryption algorithm to the destination phone.
Example of the RTP encryption algorithm carried in the SDP of the INVITE message:
m=audio 11780 RTP/SAVP 0 8 18 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80
502