ZyXEL Communications VMG8825-B Series, Руководство пользователя, Страница: 238 / 360

Chapter 23 Voice
VMG8825-B Series User’s Guide
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Use DHCP
Option 120 First Select this to enable the SIP server via DHCP option 120.
RTP Port Range
Start Port
End Port
Enter the listening port number(s) for RTP traffic, if your VoIP service provider gave you this
information. Otherwise, keep the default values.
To enter one port number, enter the port number in the Start Port and End Port fields.
To enter a range of ports,
• enter the port number at the beginning of the range in the Start Port field.
• enter the port number at the end of the range in the End Port field.
SRTP Support
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.
The VMG 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.
Crypto Suite Select the encryption and authentication algorithm set used by the VMG 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 VMG 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.
PCM - 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.
SIP INFO - send the DTMF tones in SIP messages.
Transport Type
Transport Type Select the transport layer protocol UDP or TCP (usually UDP) used for SIP.
Ignore Direct IP Select Enable to have the connected CPE devices accept SIP requests only from the SIP proxy/
register server specified above. SIP requests sent from other IP addresses will be ignored.
FAX Option This field controls how the VMG handles fax messages.
G711 Fax
Passthrough Select this if the VMG should use G.711 to send fax messages. You have to also select which
operating codec (
G.711Mulaw or G.711Alaw ) to use for encoding/decoding FAX data. The
peer devices must use the same settings.
T38 Fax Relay Select this if the VMG should send fax messages as UDP or TCP/IP packets through IP networks.
This provides better quality, but it may have inter-operability problems. The peer devices must
also use T.38.
QoS Tag
Table 105 VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued)
LABEL DESCRIPTION