Difference between revisions of "Voice-only with different codecs"

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* From the green line can be concluded that there is approximately 9000 Bytes/second of audio traffic between SIP-phone 201 and its Trixbox. This corresponds to 9000*8/1000=72kbps audio traffic. This corresponds to 36kbps audio traffic in one direction. To calculate the overhead, we have to calculate the UDP and SIP overhead. The UDP overhead is 28 bytes/packet at 50 packets/s giving 50*28=1400 bytes/s which is 1400*8/1000 = 12 kbps.  
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* From the graph there can be seen that the red line represent more data traffic than the blue line. This means that the IAX protocol used between the Trixboxes uses less packet overhead than the SIP protocol used between the SIP-phone and the Trixbox.
 
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* Consider the red and the green line which represent traffic to and from the SIP-phone using the g711u codec and the the SIP-phone using the GSM codec respectively. One can conclude that the traffic used by the g711u codec is much more than the traffic used by the GSM protocol.  
 
 
UDP: 28 bytes/packet, 50 packets/s, giving 50*28=1400 bytes/s. SIP overhead: 
 
 
 
 
* From the spikes in the black line there can be seen that a SIP protocal is used.
 
* From the spikes in the black line there can be seen that a SIP protocal is used.
  
This does not tell us which Asterisk box makes the conversion between the GSM codec and the G711u codec.
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However, this does not tell us which Asterisk box makes the conversion between the GSM codec and the G711u codec. Therefore another graph was made with the traffic measured only in one direction.
Therefore we made another graph with the traffic measured only in one direction.
 
  
 
Used filters in I/O-graph:
 
Used filters in I/O-graph:

Revision as of 14:04, 23 February 2007

Experimental Setup

In order to gain inside in the bandwitdh usage when SIP-phones use different codecs, the following experimental setup is used. A voice-call has been established between SIP-phone 201, which uses the GSM-codec and SIP-phone 301, which uses the G711u-codec.

Schematic overview of all PC's running which software.

Results

With WireShark data has been collected and the following filters are used in the in I/O-graph:

Filter-Black: proto=sip
Filter-Red  : ip.addr == 192.168.1.3   && ip.addr == 192.168.1.106
Filter-Green: ip.addr == 192.168.1.189 && ip.addr == 192.168.1.101
Filter-Blue : ip.addr == 192.168.1.106 && ip.addr == 192.168.1.101
Filter-Pink : ip.addr == 192.168.1.100 && ip.addr == 192.168.1.101

Wireshark-graph GSM G711 traffic Bps.png

Legenda:

  • Blue: Traffic between the Asterisk-boxes.
  • Red: Traffic to and from SIP phone 301.
  • Pink: Traffic to and from SIP phone 202.
  • Green: Traffic to and from SIP phone 201.


  • From the graph there can be seen that the red line represent more data traffic than the blue line. This means that the IAX protocol used between the Trixboxes uses less packet overhead than the SIP protocol used between the SIP-phone and the Trixbox.
  • Consider the red and the green line which represent traffic to and from the SIP-phone using the g711u codec and the the SIP-phone using the GSM codec respectively. One can conclude that the traffic used by the g711u codec is much more than the traffic used by the GSM protocol.
  • From the spikes in the black line there can be seen that a SIP protocal is used.

However, this does not tell us which Asterisk box makes the conversion between the GSM codec and the G711u codec. Therefore another graph was made with the traffic measured only in one direction.

Used filters in I/O-graph:

Filter-Black: ip.src == 192.168.1.101 && ip.dst == 192.168.1.106
Filter-Red  : ip.src == 192.168.1.3   && ip.dst == 192.168.1.106
Filter-Green: ip.src == 192.168.1.189 && ip.dst == 192.168.1.101
Filter-Blue : ip.src == 192.168.1.106 && ip.dst == 192.168.1.101
Filter-Pink : ip.src == 192.168.1.100 && ip.dst == 192.168.1.101

Wireshark-graph GSM G711 traffic Bps oneway-traffic.png

Legenda:

  • Blue: Traffic between the Asterisk-boxes.
  • Red: Traffic to and from SIP phone 301.
  • Pink: Traffic to and from SIP phone 202.
  • Green: Traffic to and from SIP phone 201.

Conclusions

  • Tsah