Secure Real-Time Transport Protocol - Yealink SIP-T4X Administration Manual

2. Click Browse to locate the certificate (*.pem or *.cer) from your local system.
3. Click Upload to upload the certificate.
The dialog box pops up to prompt "Success: The Server Certificate has been
loaded! Rebooting, please wait...".
Secure Real-Time Transport Protocol (SRTP) encrypts RTP streams during VoIP phone
calls to avoid interception and eavesdropping. The parties participating in the call must
enable SRTP simultaneously. When this feature is enabled on both phones, the
encryption algorithm utilized for the session is negotiated between 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
inline:NzFlNTUwZDk2OGVlOTc3YzNkYTkwZWVkMTM1YWFj
a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:NzkyM2FjNzQ2ZDgxYjg0MzQwMGVmMGUxMzdmNWFm
a=crypto:3 F8_128_HMAC_SHA1_80 inline:NDliMWIzZGE1ZTAwZjA5ZGFhNjQ5YmEANTMzYzA0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
Configuring Security Features
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