ZyXEL VMG8324 User Guide - Page 245
Any_WAN, Multi_WAN, Start Port, End Port, AES_CM_128_HMAC_SHA1_80, AES_CM_128_HMAC_SHA1_32,
View all ZyXEL VMG8324 manuals
Add to My Manuals
Save this manual to your list of manuals |
Page 245 highlights
Chapter 21 Voice Table 114 VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued) LABEL Bound Interface Name 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. Outbound Proxy Outbound Proxy Address 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). Outbound Proxy Port Enter the SIP outbound proxy server's listening port, if your VoIP service provider gave you one. Otherwise, keep the default value. 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, SRTP Support SRTP Support • 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. Crypto Suite 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. DTMF Mode DTMF Mode 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. Transport Type Transport Type SIP INFO - send the DTMF tones in SIP messages. Select the transport layer protocol UDP or TCP (usually UDP) used for SIP. VMG8324-B10A / VMG8324-B30A Series User's Guide 245