simulating a voice call
Please re Cisco Doc 24121:
Measuring Delay,Jitter,and Packet Loss with Cisco IOS SAA and RTTMON
http://www.cisco.com/warp/public/126/saa.html
VoIP typically tolerates delays up to 150 ms(RTT) before the quality of the call is
unacceptable.
Voice call can tolerates as much as 6% packet loss before the quality of the call is
unacceptable.
Deploying Delay and Jitter Agent Routers
Where to Deploy
Delay and jitter can be measured by deploying Cisco routers 17xx or higher with Cisco IOS
software code version 12.05T or higher, and configuring the Cisco IOS SAA features. The
routers should be placed in the campus networks next to hosts. This provides statistics for end-to-end connections.
Simulating a Voice Call
One of the strengths of using SAA as the testing mechanism is that a voice call can be
simulated. For example, imagine you want to simulate a G.711 voice call. You know that it
uses RTP/UDP ports 14384 and above, it is approximately 64 kb/s, and the packet size is
200 bytes {(160 bytes of payload + 40 bytes for IP/UDP/RTP (uncompressed) }.You can
simulate that type of traffic by setting up the SAA Delay/Jitter Probe as shown below.
The jitter operation needs to do this:
Send the request to RTP/UDP port number 14384.
Send 172 byte packets (160 payload + 12 byte RTP header size) + 28 bytes (IP + UDP).
Send 3000 packets for each frequency cycle.
(it takes 20ms to send one packet.In one second,you can send 50 packets.To simulate one
minute call,you have to send 3000 packets.got it?)
Send every packet 20 milliseconds apart for a duration of 60 seconds and sleep 10 seconds before starting the next frequency cycle.
Those parameters give 64 kb/s for 60 seconds.
((3000 datagrams * 160 bytes per datagram)/ 60 seconds)) * 8 bits per byte = 64 kb/s
The configuration on the router appears as follows:
rtr 1
type jitter dest-ipaddr 10.2.2.20 dest-port 14384 source-ipaddr 10.14.2.8 num-packets 3000
request-data-size 172
tos 184 (DSCP 46)
frequency 70
rtr schedule 1 life forever start-time now
Note: IP+UDP is not considered in the request-data-size as the router adds them automatically to the size internally.
On 12.4+,the configuration appears as follows:
ip sla monitor 1
type jitter dest-ipaddr 10.2.2.17 dest-port 14384 source-ipaddr 10.14.2.1 codec g711alaw
codec-numpackets 3000
tos 184 (DSCP 46x4)
frequency 70
ip sla monitor schedule 1 life forever start-time now
Note:you need to enable "rtr responder" or "ip sla monitor responder" on the remote router.
Sample Data Collections
The data can be viewed using the Cisco IOS show command at the command line on the delay and jitter probes.
R1#sh rtr collection-statistics 1
On Cisco IOS 12.4:
IDL286312#sh ip sla monitor collection-statistics 1
It only shows two hours stats.
Measuring Delay,Jitter,and Packet Loss with Cisco IOS SAA and RTTMON
http://www.cisco.com/warp/public/126/saa.html
VoIP typically tolerates delays up to 150 ms(RTT) before the quality of the call is
unacceptable.
Voice call can tolerates as much as 6% packet loss before the quality of the call is
unacceptable.
Deploying Delay and Jitter Agent Routers
Where to Deploy
Delay and jitter can be measured by deploying Cisco routers 17xx or higher with Cisco IOS
software code version 12.05T or higher, and configuring the Cisco IOS SAA features. The
routers should be placed in the campus networks next to hosts. This provides statistics for end-to-end connections.
Simulating a Voice Call
One of the strengths of using SAA as the testing mechanism is that a voice call can be
simulated. For example, imagine you want to simulate a G.711 voice call. You know that it
uses RTP/UDP ports 14384 and above, it is approximately 64 kb/s, and the packet size is
200 bytes {(160 bytes of payload + 40 bytes for IP/UDP/RTP (uncompressed) }.You can
simulate that type of traffic by setting up the SAA Delay/Jitter Probe as shown below.
The jitter operation needs to do this:
Send the request to RTP/UDP port number 14384.
Send 172 byte packets (160 payload + 12 byte RTP header size) + 28 bytes (IP + UDP).
Send 3000 packets for each frequency cycle.
(it takes 20ms to send one packet.In one second,you can send 50 packets.To simulate one
minute call,you have to send 3000 packets.got it?)
Send every packet 20 milliseconds apart for a duration of 60 seconds and sleep 10 seconds before starting the next frequency cycle.
Those parameters give 64 kb/s for 60 seconds.
((3000 datagrams * 160 bytes per datagram)/ 60 seconds)) * 8 bits per byte = 64 kb/s
The configuration on the router appears as follows:
rtr 1
type jitter dest-ipaddr 10.2.2.20 dest-port 14384 source-ipaddr 10.14.2.8 num-packets 3000
request-data-size 172
tos 184 (DSCP 46)
frequency 70
rtr schedule 1 life forever start-time now
Note: IP+UDP is not considered in the request-data-size as the router adds them automatically to the size internally.
On 12.4+,the configuration appears as follows:
ip sla monitor 1
type jitter dest-ipaddr 10.2.2.17 dest-port 14384 source-ipaddr 10.14.2.1 codec g711alaw
codec-numpackets 3000
tos 184 (DSCP 46x4)
frequency 70
ip sla monitor schedule 1 life forever start-time now
Note:you need to enable "rtr responder" or "ip sla monitor responder" on the remote router.
Sample Data Collections
The data can be viewed using the Cisco IOS show command at the command line on the delay and jitter probes.
R1#sh rtr collection-statistics 1
On Cisco IOS 12.4:
IDL286312#sh ip sla monitor collection-statistics 1
It only shows two hours stats.
posted by Jun Wen at 4:12 PM
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