Thanks Gordon, but I would like how can I calculate for myself... The 
HCA is a software already done!

Just I would like to know how can I calculate delay from that:

TIMESTAMP
18:31:01.248701000
SENDER REPORT
MSW: 3351349861
LSW: 1073741824
Sender's packet count: 2
Sender's octet count: 320
RECEIVER REPORT
Cumulative number of packets lost: 0
Extended highest sequence number received: 0
Interarrival jitter: 0
Last SR timestamp: 2321891328
Delay since last SR timestamp: 1

TIMESTAMP
18:31:01.280097000
SENDER REPORT
MSW: 3351349861
LSW: 1206885810
Sender's packet count: 4
Sender's octet count: 640
RECEIVER REPORT
Cumulative number of packets lost: 0
Extended highest sequence number received: 0
Interarrival jitter: 0
Last SR timestamp: 2321893359
Delay since last SR timestamp: 1

TIMESTAMP
18:31:06.280193000
SENDER REPORT
MSW: 3351349866
LSW: 1541893259
Sender's packet count: 190
Sender's octet count: 30400
RECEIVER REPORT
Cumulative number of packets lost: 0
Extended highest sequence number received: 58034
Interarrival jitter: 165
Last SR timestamp: 2322226151
Delay since last SR timestamp: 1

Regards,
David



En/na Beith, Gordon ha escrit:
> You can have a look at our HCA (Hammer Call Analyzer) product, which
> calculates a lot of these metrics for you.
> http://www.empirix.com/ecd/ecforms/process/hca-process.asp
> Regards,
>       Gordon
>
> -----Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of David
> Grau Serra
> Sent: Thursday, March 23, 2006 11:23 AM
> To: [email protected]
> Subject: Re: [Sip-implementors] SIP: QoS
>
> Hello all,
>
> I am sending again my email, does anybody know any rtcp mailing list?
> I am working with x-lite softphone.
>
>   
>> Hi all list,
>>
>> My presentation: I am david grau and I am from barcelona. Excuses for
>>     
> my 
>   
>> poor english.
>>
>> I don't know exactly if this mail-list is the appropiate site to say 
>> what I am doing... so, if not, please, tell me where can I go to say
>>     
> that...
>   
>> My project objective is monitor and evaluate a wireless VoIP (SIP)
>>     
> peer 
>   
>> to peer session by analysing RTCP control stream and filtering 
>> performance parameters such as jitter, delay and packet loss. These 
>> values will be dynamically plotted in order to provide a visual
>>     
> synopsis 
>   
>> of network health.
>> I use a command line network sniffer to capture the relevant control 
>> packets i.e. RTCP sender and receiver reports.
>> I parse the RTCP sender and receiver reports to filter out values for 
>> jitter, delay and packet loss.
>>
>> I show you part of the parsing:
>>
>> TIMESTAMP
>> 18:31:01.248701000
>> SENDER REPORT
>> MSW: 3351349861
>> LSW: 1073741824
>> Sender's packet count: 2
>> Sender's octet count: 320
>> RECEIVER REPORT
>> Cumulative number of packets lost: 0
>> Extended highest sequence number received: 0
>> Interarrival jitter: 0
>> Last SR timestamp: 2321891328
>> Delay since last SR timestamp: 1
>>
>> TIMESTAMP
>> 18:31:01.280097000
>> SENDER REPORT
>> MSW: 3351349861
>> LSW: 1206885810
>> Sender's packet count: 4
>> Sender's octet count: 640
>> RECEIVER REPORT
>> Cumulative number of packets lost: 0
>> Extended highest sequence number received: 0
>> Interarrival jitter: 0
>> Last SR timestamp: 2321893359
>> Delay since last SR timestamp: 1
>>
>> TIMESTAMP
>> 18:31:06.280193000
>> SENDER REPORT
>> MSW: 3351349866
>> LSW: 1541893259
>> Sender's packet count: 190
>> Sender's octet count: 30400
>> RECEIVER REPORT
>> Cumulative number of packets lost: 0
>> Extended highest sequence number received: 58034
>> Interarrival jitter: 165
>> Last SR timestamp: 2322226151
>> Delay since last SR timestamp: 1
>>
>>
>> I am a newbie and I don't know what exactly I have to do with these 
>> values. I need to calculate delay, jitter and packet lost but I am 
>> confuse...
>> Just I know how can I calculate jitter: Jitter is measured by RTCP 
>> software and included in the RR messages sent by the receiver. As this
>>     
>
>   
>> value is measured in sampling units, in order to convert to time
>>     
> units, 
>   
>> one must divide by the sampling rate of the media codec. Again using
>>     
> the 
>   
>> example data, we have jitter values of 227 and 335 sampling units, 
>> which, when divided by ITU-T G.711 sampling rate of 8 kHz, correspond
>>     
> to 
>   
>> 28.375 ms and 44.125 ms, respectively.
>>
>> Is anybody available to explain an example to calculate delay and
>>     
> packet 
>   
>> lost?
>> Thanks in advance.
>>
>> david grau
>>
>> _______________________________________________
>> Sip-implementors mailing list
>> [email protected]
>> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>>
>>
>>   
>>     
> _______________________________________________
> Sip-implementors mailing list
> [email protected]
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
>
>   
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