Hi Jose,
I looked at ethereal but I don't think it's possible to use that in real time i.e. you have to start the capture and when you think the call is over stop the capture and examine the results. I would like something in real-time…that makes me think tethereal perhaps might be an option….
Thanks,
Aisling.
-----Original Message-----
From: Voipers Portugal [mailto: voipers@gmail.com]
Sent: 30 January 2006 11:47
To: Aisling
Cc: John Todd; serusers@iptel.org; clona@cyberhouse.no
Subject: Re: [Serusers] SER RTP QoS Monitoring
Ethereal does that automatically, so I think you can start there. As for RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t when I tested a few months ago).
Jose Simoes
On 1/30/06, Aisling < ashling.odriscoll@cit.ie> wrote:
Hi,
Thanks to John, Nils and Atle for the replies. So if I am to proceed
with extracting QoS info from RTCP it is now my understanding that I
would either have to have my QoS gathering application at:
1) the media proxy (used primarily for nat situations)
2) where the voice is going peer to peer, it would need to be placed at
the end points. However what I'm wondering about here is would it be
possible to have a QoS collector application running on another machine
on the same network as the two endpoints (obviously presuming they're
all on the same network) which could intercept the sender and receiver
reports? And if so does anyone have any idea if there's another way to
distinguish RTCP packets without knowing the IP address or port?
Many thanks,
Aisling.
-----Original Message-----
From: John Todd [mailto: jtodd@loligo.com]
Sent: 29 January 2006 22:33
To: Aisling O'Driscoll; serusers@iptel.org
Subject: Re: [Serusers] SER RTP QoS Monitoring
Hello -
Your method #1 (RTCP) is the best solution, it would seem. Method
#2 (INFO) is sort-of supported in some devices, like the Cisco 79xx
series with SIP software - they support insertion of some basic call
element data into the BYE message. Method #3 is complex, as it would
require the SNMP collector to have full knowledge of all call states
on the devices that were queried (though possibly this is not more
complex than the RTCP collection process, either.) In any of these
three cases, SER does not collect the call quality data - the
endpoints or media proxies are the only places that see this data.
If you run a mediaproxy in conjunction with SER, then it may be
possible to collect and report on that data, but there are drawbacks
to a design of that nature.
For the long term, I would suggest taking a look at this draft,
which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to
be transmitted via SIP messages between media-handling entities and
other reporting systems:
http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-
08.txt
While we are still a while away from seeing vendor (or open source)
implementations, this seems like a reasonable proposal. It allows
for SIP-based publication or subscription of call quality events by
various network elements, and it will permit arbitrary querying of
this data at any time interval (not just at the end of the call.)
JT
>Hi,
>
>I want to be able to extract QoS parameters such as jitter, delay and
>packet loss from VoIP calls made using my SER(both peer to peer for
>my public clients and also for clients where the voice will be routed
>with mediaproxy). I would like to develop something as opposed to
>purchasing a software package.
>
>>From searching the Internet the last few days I have seen that the
>following options seem to be available:
>
>1) RTCP
>RTCP stream will contain this information. The problem is that
>sniffers cannot clearly distinguish RTCP packets from other UDP
>packets unless something like an IP address or specific UDP port is
>supplied. I had considered using TCPDUMP but then where do I position
>this in the network when voice is going peer to peer between client
>and I'm not sure how this can work to monitor all calls when I might
>not be aware of the end user IP addresses.
>
>2) SIP INFO extension
>This also seem to be an option but most off the shelf phones don't
>seem to support this and this would required modification of a SIP ua.
>
>3) SNMP
>I thought maybe SNMP might be an option but the SER snmp module no
>longer exists...
>
>Does anyone have any comments on the above? Are the statements that I
>made correct or can anyone think of other ways to monitor the voice
>QoS? I am trying to understand how commercial applications have
>accomplished this.
>
>Many thanks in advance for advice.
>Aisling.
>
>>---- Original Message ----
>>From: clona@cyberhouse.no
>>To: ashling.odriscoll@cit.ie
>>Subject: Re: [Serusers] Status of SER SNMP module
>>Date: Sun, 29 Jan 2006 20:30:02 +0100
>>
>>>
>>>Hi Aisling,
>>>
>>>The SNMP module is dead as a killed turtle :(
>>>
>>>-Atle
>>>* Aisling O'Driscoll < ashling.odriscoll@cit.ie> [060129 20:26]:
>>>> Hello,
>>>>
>>>> I'm just wondering what the current status of the SER SNMP module
>>is?
>>>> Is it currently supported and is there somewhere I can find
>>>> documentation on it?
>>>>
>>>> Many thanks,
> >>> Aisling.
>
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