Avaya 4600 Series 사용자 설명서

다운로드
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4600 Series IP Telephone LAN Administrator’s Guide
QoS
4-22
encodings that do not terminate in 000. The specific treatment intended by these custom DSCPs 
will not necessarily be carried out by routers outside the customer’s own network.
To support DiffServ, the 4600 Series IP Telephones are administered at the switch in accordance 
with the documentation for the switch.
UDP Port Selection
4
Some data networks include equipment that can perform UDP port selection. This is a mechanism 
by which packets with port numbers in a given range are given priority over packets with port 
numbers outside that range.
To support UDP port selection, the 4600 Series IP Telephones can be administered either from the 
network, via appropriate administration of the DHCP or TFTP servers, or at the telephone itself, via 
dialpad input. Specific implementation details for local administration are in the 4600 Series IP 
Telephone Installation Guide
, and, for remote administration, in 4600 Series IP Telephone Scripts 
and Application Files
 on page 4-18. In summary, the system value MCPORT represents the port 
on the TN2302AP board. This port number can be used to administer routers, etc. that support 
UDP port selection, to maximize the priority given to the voice packets being exchanged between 
the PBX and the telephone.
The default value for MCPORT is 1719. The switch must be administered to use a port within the 
proper range for the specific LAN, and the IP Telephone(s) will copy that port. A related parameter 
is PORTAUD, which is the RTP port used by the switch. In accordance with standards RFC 1889 
and 1890, the IP Telephone uses a default value for PORTAUD of 5004. MCPORT and PORTAUD 
are both administrable (see Administering Options for the 4600 Series IP Telephones on page 
4-24
).
QoS with 4620 and 4630 IP Telephones
4
As of Release 1.7, both the 4620 and 4630 telephones offer the end user an opportunity to monitor 
network audio performance while on a call. The user guides for each phone go into specific details 
on how the user gets to the appropriate screen; this document tells you what the end user can see, 
and what it means. Specifically, the following parameters are displayed in real-time to users on the 
appropriate screens, while on a call:
Parameter
Possible Values
Audio Connection Present?
Yes (if a receive RTP stream has been established)
No (if a receive RTP stream has not been established)
Received Audio Coding 
G.711 or G.729
Silence Suppression
Yes (if the telephone knows the far-end has silence 
suppression Enabled)
No (if the telephone knows the far-end has silence 
suppression Disabled, or the telephone does not know 
either way)