Voice-only with different codecs

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.

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

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 green line can be concluded that there is approximately 10000 Bytes/second of audio traffic between SIP-phone 201 and its Trixbox. This corresponds to 10000*8/1000=80kbps audio traffic. This corresponds to 40kbps audio traffic in one direction.
• 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. 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

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