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Administrator's Manual 4. Web-based Management

Version 1.0.2 51 October 2009

4.6.3 Configuring the Media Streaming Parameters

The media streaming parameters are configured in the ‘Media Streaming’ page, as described

below.

¾ To define the media streaming parameters:

1. Access the ‘Media Streaming’ page (Configuration tab > Voice Over IP menu > Media

Streaming).

Figure 4-10: Media Streaming Page

2. Under the Media Streaming Parameters group, configure the following parameters:

• RTP Port Range: Defines the port range for Real Time Protocol (RTP) voice transport.

• DTMF Relay RFC 2833 Payload Type: The RTP payload type used for RFC 2833

DTMF relay packets. Range is 0 to 255. The default is 101.

3. Under the Quality of Service Parameters group, in the Type of Service (ToS) field, enter

the quality of service in hexadecimal format. This is a part of the IP header that defines the

type of routing service to be used to tag outgoing voice packets, originated from the phone. It

is used to tell routers along the way that this packet should get specific QoS. Leave this

value as 0xb8 (default) if you are unfamiliar with the Differentiated Services IP protocol

parameter.

4. Under the Codecs group, select the codec(s) and packetization time that you want the

phone to use. You can select up to five codecs, where the first codec in the list is given the

highest priority, and the rest in descending order of appearance. To make a call, at least one

codec must be configured. In addition, for best performance it is recommended to select as

many codecs as possible.

When you start a call to a remote party, your available codecs are compared with the remote

party's to determine the codec to use. If there is no codec that both parties have made

available, the call attempt fails. Note that if more than one codec is common to both parties,

you cannot force which of the common codecs are used by the remote party's client. If you

do wish to force the use of a specific codec, configure the list with only that specific codec.

The Packetization Time is the length of the digital voice segment that each packet holds. The

default is 20 millisecond packets. Selecting 10 millisecond packets reduces the delay, but

increases the bandwidth consumption.

5. Under the G.723 Bitrate group, from the G.723 Bitrate drop-down list, select low or high bit

rate for G.723.

6. Click Submit.

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