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IOS IP SLA Voice Port Not Found

Ffound a need to confirm some voip packet loss today against what Prognosis IPT Assessor, found the IOS IP SLA config changed in 12.4.22T4, below is running-config, but I haven't figured out why Voice Port Not Found is showing up. It is because I made my port 14000 vs 16384?

SLA-Sender (config)#ip sla 3

SLA-Sender (config-ip-sla)# voip rtp 10.225.106.2 source-ip 10.225.33.249 source-voice-port 14000 codec g711ulaw duration 60 advantage-factor 0

Exit

SLA-Sender (config)#ip sla schedule 3 life 36000 start-time now

SLA-Sender#show ip sla stat

IPSLAs Latest Operation Statistics

IPSLA operation id: 3

Type of operation: rtp

Latest operation start time: *18:24:54.413 dst Thu Jul 4 2002

Latest operation return code: Voice Port Not Found

Latest RTT (milliseconds): 0

Source to Destination Path Measurements:

        Interarrival Jitter: 0

        Packets Sent: 0

        Packets Lost: 0

        Estimated R-factor: 0   MOS-CQ: 0.00

Destination to Source Path Measurements:

        Interarrival Jitter: 0

        Packets Sent: 0

        Packets Lost: 0

        Estimated R-factor: 0   MOS-CQ: 0.00

Operation time to live: 71053 sec

Operational state of entry: Active

Last time this entry was reset: Never

SLA-Sender#show ip sla configuration

IP SLAs Infrastructure Engine-II

Entry number: 3

Owner:

Tag:

Type of operation: rtp

Target address/Source address: 10.225.106.2/10.225.33.249

Source voice port: 14000

RTP session duration (seconds): 60

Codec Type: g711ulaw

Advantage Factor: 0

Operation timeout (milliseconds): 5000

Type Of Service parameters: 0x0

Vrf Name:

Schedule:

   Operation frequency (seconds): 90  (not considered if randomly scheduled)

   Next Scheduled Start Time: Start Time already passed

   Group Scheduled : FALSE

   Randomly Scheduled : FALSE

   Life (seconds): 36000

   Entry Ageout (seconds): never

   Recurring (Starting Everyday): FALSE

   Status of entry (SNMP RowStatus): Active

Distribution Statistics:

   Number of statistic hours kept: 2

   Number of statistic distribution buckets kept: 1

   Statistic distribution interval (milliseconds): 20

Threshold (milliseconds): 5000 (not considered if react RTT is configured)

SLA-Responder(config)# ip sla responder

SLA-Responder#show ip sla responder

IP SLAs Responder is: Enabled

Number of control message received: 62 Number of errors: 0

Recent sources:

        10.225.33.249 [16:13:58.165 dst Tue Jul 13 2010]

        10.225.33.249 [16:13:58.137 dst Tue Jul 13 2010]

        10.225.33.249 [16:13:58.113 dst Tue Jul 13 2010]

        10.225.33.249 [16:12:28.160 dst Tue Jul 13 2010]

        10.225.33.249 [16:12:28.132 dst Tue Jul 13 2010]

Recent error sources:

SLA-Responder# show ip sla config

!

SLA-Responder#show ip sla stat

IPSLAs Latest Operation Statistics

SLA-Responder #

-----Original Message-----
From: cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] On Behalf Of Nick Matthews
Sent: Friday, July 09, 2010 1:29 PM
To: Drew Weaver
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] Simulate SIP traffic end to end

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@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

>

>

>

> _______________________________________________

> cisco-voip mailing list

> cisco-voip@puck.nether.net

> https://puck.nether.net/mailman/listinfo/cisco-voip

>

>

_______________________________________________

cisco-voip mailing list

cisco-voip@puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

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