ZyXEL GS-105I User Guide - Page 234

VoIP, PX7511-B0 User's Guide, Table 98, VoIP > SIP > SIP Service Provider > Add

Page 234 highlights

Chapter 21 VoIP Table 98 VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued) LABEL DESCRIPTION Use DHCP Option 120 First RTP Port Range Start Port End Port Select this to enable the SIP server via DHCP option 120. 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 Zyxel 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 Zyxel 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 Zyxel 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 Ignore Direct IP FAX Option G711 Fax Passthrough T38 Fax Relay QoS Tag SIP INFO - send the DTMF tones in SIP messages. Select the transport layer protocol UDP or TCP (usually UDP) used for SIP. 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. This field controls how the Zyxel Device handles fax messages. Select this if the Zyxel Device should use G.711 to send fax messages. The peer devices must use the same settings. Select this if the Zyxel Device 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. PX7511-B0 User's Guide 234

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Chapter 21 VoIP
PX7511-B0 User’s Guide
234
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 Zyxel 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.
Crypto Suite
Select the encryption and authentication algorithm set used by the Zyxel 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.
DTMF Mode
DTMF Mode
Control how the Zyxel 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.
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 Zyxel Device handles fax messages.
G711 Fax
Passthrough
Select this if the Zyxel Device should use G.711 to send fax messages. The peer devices must use
the same settings.
T38 Fax Relay
Select this if the Zyxel Device 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 98
VoIP > SIP > SIP Service Provider > Add new provider/Edit (continued)
LABEL
DESCRIPTION