Media Stream; Figure 5-16 Media Stream Configuration Interface; Table 5-16 Media Stream Configuration Parameters - Tadiran Telecom TGW1E1-2G User Configuration Manual

Trunk gateway series
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TGWxE1-2G PRI (ISDN) Trunk
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5.5.8 Media Stream

Click the Advanced > Media Stream tab to open this interface.

Figure 5-16 Media Stream Configuration Interface

Table 5-16 Media Stream Configuration Parameters

Name
Min. RTP port
Max. RTP port
iLBC payload type
G.723.1 rate
TOS bits
Min. Jitter buffer
Max. Jitter buffer
RTP drop SID
5-31
Description
The minimum value of UDP ports for RTP transmission and receiving. The parameter
must be greater than or equal to 3000.
It is recommended that the value be equal to or greater than 10000.
Note: Each phone call will occupy RTP and RTCP ports.
The maximum values of UDP ports for RTP's transmission and receiving.
It is recommended that the value be greater than or equal to "2× number of lines + min.
RTP port".
Set the RTP payload type of iLBC. The default value is 97. Accepted range is 97-127.
The parameter shall be configured in conformity to that of platform.
It refers to the RTP payload type number that indicates the transmitted packet, which is
using the iLBC CODEC. The parameter value should be the same for both ends of the call.
Set G.723.1 coding rate.The default value is 6300. The optional parameters are as follows:
5300: the Bit rate is 5.3k per second
6300: the Bit rate is 6.3k per second
This parameter specifies the quality assurance of services with different priorities. The
factory setting is 0x0C.
For example, TOS=0xB8 indicates that the priority of the service quality is 5, with a
requirement on low delay and high throughput. There is no requirement on the reliability.
RTP Jitter Buffer is constructed to reduce the influence brought by network jitter.
The factory default value is 3.
Network jitter refers the variation in the delay of received packets that are caused by
congestion in the IP network. Thus, we need to to configure jitter buffer in order to
minimize delay variations. If set is too small it will lead to call quality degradation.
RTP Jitter Buffer helps to reduce the influence brought by network jitter.
The factory default value is 50.
Same explanation as Min. Jitter buffer, above. If set too large it will introduced additional
delay which will lead to difficulty in conversation.
Determines whether to discard received RTP SID voice packets. By default, SID voice
packets will not be dropped.
Note: RTP SID packets should be dropped only when do not conform to the specifications.
Nonstandard RTP SID data could generate noise for calls.
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