Chapter 9 Voice

 

Table 58 VoIP > SIP > SIP Service Provider (continued)

 

LABEL

DESCRIPTION

 

SRTP Support

When you make a VoIP call using SIP, the Real-time Transport Protocol

 

 

(RTP) is used to handle voice data transfer. The Secure Real-time

 

 

Transport Protocol (SRTP) is a security profile of RTP. It is designed to

 

 

provide encryption and authentication for the RTP data in both unicast

 

 

and multicast applications.

 

 

SRTP uses the Advanced Encryption Standard (AES) cipher for data

 

 

encryption. The P-2812HNU-51c supports encryption using AES with a

 

 

128-bit key. To protect data integrity, SRTP uses a Hash-based Message

 

 

Authentication Code (HMAC) calculation with Secure Hash Algorithm

 

 

(SHA)-1 to authenticate data. HMAC SHA-1 produces a 80 or 32-bit

 

 

authentication tag that is appended to the packet.

 

 

Both the caller and callee should use the same algorithms to establish

 

 

an SRTP session.

 

 

 

 

SRTP Support

Select this option to enable SRTP on the P-2812HNU-51c.

 

 

 

 

Crypto Suite

Select the encryption and authentication algorithm set used by the P-

 

 

2812HNU-51c to set up an SRTP media session with the peer device.

 

 

Select AES_CM_128_HMAC_SHA1_80 or

 

 

AES_CM_128_HMAC_SHA1_32 to enable both data encryption and

 

 

authentication for voice data.

 

 

Select AES_CM_128_NULL to use 128-bit data encryption but disable

 

 

data authentication.

 

 

Select NULL_CIPHER_HMAC_SHA1_80 to disable encryption but

 

 

require authentication using the default 80-bit tag.

 

 

 

 

DTMF Mode

 

 

 

 

 

DTMF Mode

Control how the P-2812HNU-51c handles the tones that your telephone

 

 

makes when you push its buttons. You should use the same mode your

 

 

VoIP service provider uses.

 

 

RFC2833 - send the DTMF tones in RTP packets.

 

 

InBand - send the DTMF tones in the voice data stream. This method

 

 

works best when you are using a codec that does not use compression

 

 

(like G.711). Codecs that use compression (like G.729 and G.726) can

 

 

distort the tones.

 

 

SIPInfo - send the DTMF tones in SIP messages.

 

 

 

 

Transport Type

 

 

 

 

 

Transport Type

Select the transport layer protocol (TCP, UDP or TLS) used for SIP.

 

 

AUTO is available when you select the Support Locating SIP Server

 

 

option. If you select AUTO here, the P-2812HNU-51c sends a DNS

 

 

Name Authority Pointer (NAPTR) query to locate the SIP server and get

 

 

the supported transport layer protocol(s).

 

 

 

 

FAX Option

This field controls how the P-2812HNU-51c handles fax messages.

 

 

 

 

G.711 Fax

Select this if the P-2812HNU-51c should use G.711 to send fax

 

Passthrough

messages. The peer devices must also use G.711.

 

 

 

 

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P-2812HNU-51c User’s Guide