Voice-only with different codecs - Revision history

Rubzzzz: /* Conclusions */

2007-02-23T13:18:45Z

Conclusions

2007-02-23T13:15:00Z

Results

2007-02-23T13:04:45Z

Results

2007-02-23T12:53:40Z

Results

2007-02-23T12:50:46Z

Results

2007-02-23T12:28:34Z

Results

2007-02-23T12:28:14Z

Results

2007-02-23T12:27:51Z

Results

2007-02-23T12:24:20Z

Results

2007-02-23T12:22:59Z

Results

← Older revision		Revision as of 13:18, 23 February 2007
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	== Conclusions ==		== Conclusions ==

−	* ~~Tsah~~	+	* 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.

@@ Line 43: / Line 43: @@
 * '''Pink''': Traffic to and from SIP phone 202.
 * '''Green''': Traffic to and from SIP phone 201.
 == Conclusions ==
 * Tsah

@@ Line 23: / Line 23: @@
-* 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.
+* 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.
-This does not tell us which Asterisk box makes the conversion between the GSM codec and the G711u codec.
+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:

← Older revision		Revision as of 12:53, 23 February 2007
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−	* 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. 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 5028=1400 bytes/s.	+	* 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 90008/1000=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 5028=1400 bytes/s which is 1400*8/1000 = 12 kbps.

← Older revision		Revision as of 12:50, 23 February 2007
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−	* 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 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 100008/1000=80kbps audio traffic. This corresponds to 40kbps 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 5028=1400 bytes/s.
		+
		+
		+	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.

← Older revision		Revision as of 12:22, 23 February 2007
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	[[image:Wireshark-graph_GSM_G711_traffic_Bps.png]]		[[image: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 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.