Voice-only with different codecs

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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.

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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
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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
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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.
  • In the above graph there can be seen that the amount of traffic between the SIP-phone using the GSM codec and its Trixbox is significantly lower than the traffic between the two Trixboxes and the traffic between the Trixbox and the SIP-phone using g711u. So the conversion between the GSM and the g711u codec is done at the Trixbox at which the SIP-phone using the GSM codec is connected.

Conclusions

  • IAX overhead is less than SIP packet overhead.
  • If two SIP-phones are connected to each other while using other codecs, the Trixbox at which the conversion between the codecs is done depends on the bandwidth set in de IAX.conf. If this is low, IAX traffic will use the lowest bandwidth codec and the other way around if the bandwidth is set high.