Bizfon 2000 Mode D'Emploi

Bizfon Manual II: Administrator's Guide  

Administrator’s Menus 

Bizfon2000 and Bizfon4000 (SW Version 4.1.x) 

 

 

The  RTP Statistics page provides detailed information about the 
established call is provided. When Bizfon serves as an RTP proxy, 
this page displays two groups (legs) of RTP statistics. For example, 
when calling from an IP Phone attached to the Bizfon’s IP line to an 
external SIP destination or from one external SIP destination to 
another through the Bizfon’s Auto Attendant. Each group of 
parameters describes characteristics of a piece of RTP stream 
composing an overall SIP session. Normally, one leg describes the 
RTP stream from caller to the Bizfon and the other leg describes the 
RTP stream from Bizfon to the destination. 

Quality - estimated call quality, which depends on RTP statistic. 
Below is the legend for Call Quality definitions on the displayed RTP 
Statistics: 

excellent – RX Lost Packets < 1%  &  RX Jitter < 20  
good - RX Lost Packets < 5%  &  RX Jitter < 80 
satisfactory - RX Lost Packets < 10%  &  RX Jitter < 150 
bad - RX Lost Packets < 20%  &  RX Jitter < 200 
very bad - RX Lost Packets > 20%  or  RX Jitter > 200 

Fig. II-99: RTP Statistics page 

Rx/Tx Codec - codec for received and transmitted RTP stream respectively. 

Rx/Tx Packets - number of RTP packets received and transmitted respectively. 

Rx/Tx Packet Size - size of RTP packet (payload) received and transmitted respectively. 

Rx Lost Packets - number of lost RTP packets for received stream. 

Rx Jitter - inter-arrival jitter is an estimate of the statistical variance of the RTP data packet inter-arrival time, measured in timestamp units. 

The inter-arrival jitter is defined to be the mean deviation (smoothed absolute value) of the difference D in packet spacing at the receiver compared 
to the sender for a pair of packets. If Si is the RTP timestamp from packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then for 
two packets i and j, D may be expressed as: 

         D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si) 

         J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16,  where J(i) is Rx Jitter for packet i. 

For more details about Jitter calculations, please refer to the RFC1889. 

Rx Maximum Delay - maximum variance (absolute value) of actual arrival time of the RTP data packet compared to estimated arrival time, 
measured in milliseconds.  

If Si is the RTP timestamp from packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then variance for packet i may be 
expressed as following:  V(i) = |(Ri - R1) - (Si - S1)| = |(Ri - Si) - (R1 - S1)|  

Rx Maximum Delay = max V(i) / 8 

RX Delay Increase Count – indicates the number of times the delay in jitter buffer is increased during the call. 

RX Delay Decrease Count - indicates the number of times the delay in jitter buffer is decreased during the call. 

 RTP Statistics is logged only when at least one of the call endpoints is located on the Bizfon. For example, it will not be logged when: 

• 

calls incoming from or addressed to the IP lines or remote extension, 

• 

calls from an external user are routed to another external user through Bizfon’s routing rules. 

In the first case, RTP statistics will be logged if remote extension or IP line user is calling locally to the Bizfon’s extension or auto attendant. 
 
The  Configure Call Quality Event Notification link leads to the Configure Call Quality Event Notification page where call quality control 
notification specifics can be configured.