Voice-only with different codecs

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Revision as of 13:11, 23 February 2007 by Rubzzzz (talk | contribs) (→‎Results)
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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

This does not tell us which Asterisk box makes the conversion between the GSM codec and the G711u codec. Therefore we made another graph 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