[Serusers] SER RTP QoS Monitoring

[John Todd]

Aisling -
   To address your points in order:

1) It would be possible to intercept the data (RTCP) QoS packets in a 
media proxy.  Of course, if you want to check QoS on all your calls, 
then all media must pass through this media proxy.  If you come up 
with some interesting way to perform this in any open-source manner, 
I'm sure the community at large would be very interested in hearing 
about it.  I believe some Session Border Controllers have this 
functionality (perhaps those vendors can chime in via private mail to 
you if they have such functionality.)

2) Peer to Peer voice is more or less impossible to capture in 
real-world situations without a media proxy.  I don't want to make 
this a rehash on why Ethernet sniffing is not viable as a production 
method, but a mesh of ethernet sniffers on a switched network would 
be exceedingly difficult, expensive, and sub-optimal.  It would 
probably be cheaper to build your own SIP endpoint devices or 
software that did QoS reporting in a centralized way, or just force 
the use of a media proxy.
    If you are trying just to intercept a single conversation, then 
Ethereal will work fine if you have access to all of the traffic on 
an unswitched local segment.

JT



At 9:36 AM +0000 1/30/06, Aisling 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 at loligo.com]
>Sent: 29 January 2006 22:33
>To: Aisling O'Driscoll; serusers at lists.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 at cyberhouse.no
>>>To: ashling.odriscoll at 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 at 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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>>serusers at lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>
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