[cisco-voip] Simulate SIP traffic end to end

[Nick Matthews]

SIP has no inherent quality - it's like asking the quality of SCCP.
You're really wanting to know about UDP generation/reception.  There
is a feature on the routers called IP SLA, and one of the options is
sending voice quality.  It will send RTP packets and give you the MOS
scores and packet loss between two routers.  There are also numerous
third party applications that do the same thing, but probably a lot
prettier.

Here's a sample config/show I did a while back in a test network:


ending side:

ip sla monitor 3
 type jitter dest-ipaddr 172.16.1.2 dest-port 16384 codec g711ulaw
codec-numpackets 30
 tos 184
 frequency 10
ip sla monitor schedule 3 start-time now


ip sla monitor 4
 type jitter dest-ipaddr 172.16.35.5 dest-port 16386 codec g711ulaw
codec-numpackets 20
 tos 184
 frequency 5
ip sla monitor schedule 4 start-time now

Receiving side:

ip sla monitor responder



R3#sh ip sla moni stat
Round trip time (RTT)   Index 3
        Latest RTT: 54 ms
Latest operation start time: 00:11:46.823 UTC Tue Jan 1 2008
Latest operation return code: OK
RTT Values
        Number Of RTT: 13
        RTT Min/Avg/Max: 108/124/153 ms
Latency one-way time milliseconds
        Number of one-way Samples: 0
        Source to Destination one way Min/Avg/Max: 0/0/0 ms
        Destination to Source one way Min/Avg/Max: 0/0/0 ms
Jitter time milliseconds
        Number of Jitter Samples: 7
        Source to Destination Jitter Min/Avg/Max: 5/6/7 ms
        Destination to Source Jitter Min/Avg/Max: 1/1/1 ms
Packet Loss Values
        Loss Source to Destination: 17          Loss Destination to Source: 0
        Out Of Sequence: 0      Tail Drop: 0    Packet Late Arrival: 0
Voice Score Values
        Calculated Planning Impairment Factor (ICPIF): 25
        Mean Opinion Score (MOS): 3.50
Number of successes: 20
Number of failures: 0
Operation time to live: 3393 sec

Round trip time (RTT)   Index 4
        Latest RTT: 2 ms
Latest operation start time: 00:12:01.167 UTC Tue Jan 1 2008
Latest operation return code: OK
RTT Values
        Number Of RTT: 20
        RTT Min/Avg/Max: 2/2/2 ms
Latency one-way time milliseconds
        Number of one-way Samples: 0
        Source to Destination one way Min/Avg/Max: 0/0/0 ms
        Destination to Source one way Min/Avg/Max: 0/0/0 ms
Jitter time milliseconds
        Number of Jitter Samples: 19
        Source to Destination Jitter Min/Avg/Max: 0/0/0 ms
        Destination to Source Jitter Min/Avg/Max: 0/0/0 ms
Packet Loss Values
        Loss Source to Destination: 0           Loss Destination to Source: 0
        Out Of Sequence: 0      Tail Drop: 0    Packet Late Arrival: 0
Voice Score Values
        Calculated Planning Impairment Factor (ICPIF): 1
        Mean Opinion Score (MOS): 4.34
Number of successes: 6
Number of failures: 0
Operation time to live: 3570 sec


On Fri, Jul 9, 2010 at 8:03 AM, Drew Weaver <drew.weaver at thenap.com> wrote:
> Is there any way to simulate SIP traffic between two hosts with some
> statistics about the quality of the network between the two points?
>
>
>
> Sort of like iperf, except with SIP traffic?
>
>
>
> thanks,
>
> -Drew
>
>
>
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