Avaya 9600 manual SIP Station Number Portability, Parameters in Real-Time Possible Values

Network Requirements

Network Audio Quality Display on 9600 Series SIP IP Telephones

All 9600 Series SIP IP Telephones give the user an opportunity to monitor network audio performance while on a call. For more information, see the telephone user guide.

While on a call, the telephones display network audio quality parameters in real-time, as shown

in Table 4:

Table 4: Parameters in Real-Time

Parameter	Possible Values
Received Audio Coding	G.711, G.722, G.726A, or G.729.
Packet Loss	No data or a percentage. Late and out-of-sequence packets are
	counted as lost if they are discarded. Packets are not counted
	as lost until a subsequent packet is received and the loss
	confirmed by the RTP sequence number.
Packetization Delay	No data or an integer number of milliseconds. The number
	reflects the amount of audio data in each RTP packet.
One-way Network Delay	No data or an integer number of milliseconds. The number is
	one-half the value RTCP or SRTCP computes for the round-trip
	delay.
Network Jitter	No data or an integer number of milliseconds reporting the
Compensation Delay	average delay introduced by the jitter buffer of the telephone.

The implication for LAN administration depends on the values the user reports and the specific nature of your LAN, like topology, loading, and QoS administration. This information gives the user an idea of how network conditions affect the audio quality of the current call. Avaya assumes you have more detailed tools available for LAN troubleshooting.

SIP Station Number Portability

The 9600 Series SIP IP Telephones provide station number portability. On startup or a reboot, the telephone attempts to establish communication with its home Personal Profile Manager (PPM)/SIP Enablement Services (SES) server based on the User Name and Password.

Assume a situation where the company has multiple locations in London and New York, all sharing a corporate IP network. Users want to take their telephone functionality from their offices in London to their New York office. When users start up their telephones in the new location and enter their credentials, the local SES/PPM server usually routes them to the local call server. With proper administration of the local SES/PPM server, the telephone knows to try its home SES/PPM server, the one in London. The user can then be automatically registered with the London SES/PPM server.

30 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0