Securing Voice Traffic With Srtp; Configuring Voice Codecs - Cisco SPA301 Administration Manual

Configuring Security, Quality, and Network Features

Configuring Voice Codecs

Configuring Voice Codecs
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide
• TLS Handshake Protocol--authenticates the server and client, and
negotiates the encryption algorithm and cryptographic keys before the
application protocol transmits or receives data.
Cisco SPA IP phones use UDP as a standard for SIP transport, but they also
support SIP over TLS for added security.
To enable TLS for the phone, navigate to Admin Login > advanced > Voice >
Ext_n. Under SIP Settings, select TLS from the SIP Transport list.

Securing Voice Traffic with SRTP

Secure Real-Time Transport Protocol (SRTP) is a secure protocol for transporting
real-time data over networks. It provides media encryption to ensure that media
streams between devices are secure and that only the intended devices receive
and read the data. Cisco SPA IP phones use SRTP to securely send and receive
voice traffic to and from phones and gateways that support SRTP. (Security
Description (RFC-4568) is supported.)
When a call is secured with SRTP, the voice conversation is encrypted so that
others cannot eavesdrop on the conversation. To enable this feature, Cisco SPA IP
phones must have a mini-certificate installed.
Defaults to prefer to use encrypted media (voice codecs). Audio packets in both
directions of outbound calls are encrypted by using SRTP.
A codec resource is considered allocated if it has been included in the SDP codec
list of an active call, even though it eventually might not be chosen for the
connection. If the G.729a codec is enabled and included in the codec list, that
resource is tied up until the end of the call whether or not the call actually uses
G.729a. If the G729a resource is already allocated (and since only one G.729a
resource is allowed per IP phone), no other low-bit-rate codec can be allocated for
subsequent calls. The only choices are G711a and G711u.
Since two G.723. 1 /G.726 resources are available per IP phone, you should disable
the use of G.729a to guarantee support for two simultaneous G.723/G.726
codecs.
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