Configuring Security Features
189
The web user interface pops up the dialog box to prompt ―Rebooting, please
wait…‖.
Secure Real-Time Transport Protocol (SRTP) provides means of encrypting the RTP
streams during VoIP phone calls to avoid interception and eavesdropping. The parties
participating in the call should enable the SRTP feature
simultaneously
. When this
feature is enabled on both phones, the IP phone will negotiate with the other phone
what type of encryption to utilize for the session. 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.
The sample of the RTP encryption algorithm carried in the SDP of the INVITE message
for reference:
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
The callee receives the INVITE message with the RTP encryption algorithm. The callee
answers the call and responses with a 200 OK message carrying the negotiated RTP
encryption algorithm.
Summary of Contents for SIP-T38G
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