ZyXEL Communications VMG892-B10A User Manual page 245

Table 113 VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued)
LABEL
Bound
Interface Name
Outbound Proxy
Outbound
Proxy Address
Outbound
Proxy Port
RTP Port Range
Start Port
End Port
SRTP Support
SRTP Support
Crypto Suite
DTMF Mode
DTMF Mode
Transport Type
Transport Type
VMG8924-B10A and VMG8924-B30A Series User's Guide
DESCRIPTION
If you select LAN or Any_WAN, the Device automatically activates the VoIP service when
any LAN or WAN connection is up.
If you select Multi_WAN, you also need to select two or more pre-configured WAN
interfaces. The VoIP service is activated only when one of the selected WAN connections is
up.
Enter the IP address or domain name of the SIP outbound proxy server if your VoIP service
provider has a SIP outbound server to handle voice calls. This allows the Device to work
with any type of NAT router and eliminates the need for STUN or a SIP ALG. Turn off any SIP
ALG on a NAT router in front of the Device to keep it from re-translating the IP address
(since this is already handled by the outbound proxy server).
Enter the SIP outbound proxy server's listening port, if your VoIP service provider gave you
one. Otherwise, keep the default value.
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.
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 Device 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.
Select the encryption and authentication algorithm set used by the Device 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.
Control how the Device 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.
Select the transport layer protocol UDP or TCP (usually UDP) used for SIP.
Chapter 21 Voice
245