<div>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 didnīt when I tested a few months ago).</div>
<div> </div>
<div>Jose Simoes<br><br> </div>
<div><span class="gmail_quote">On 1/30/06, <b class="gmail_sendername">Aisling</b> <<a href="mailto:ashling.odriscoll@cit.ie">ashling.odriscoll@cit.ie</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>Hi,<br><br>Thanks to John, Nils and Atle for the replies. So if I am to proceed<br>with extracting QoS info from RTCP it is now my understanding that I
<br>would either have to have my QoS gathering application at:<br><br>1) the media proxy (used primarily for nat situations)<br>2) where the voice is going peer to peer, it would need to be placed at<br>the end points. However what I'm wondering about here is would it be
<br>possible to have a QoS collector application running on another machine<br>on the same network as the two endpoints (obviously presuming they're<br>all on the same network) which could intercept the sender and receiver
<br>reports? And if so does anyone have any idea if there's another way to<br>distinguish RTCP packets without knowing the IP address or port?<br><br>Many thanks,<br>Aisling.<br><br>-----Original Message-----<br>From: John Todd [mailto:
<a href="mailto:jtodd@loligo.com">jtodd@loligo.com</a>]<br>Sent: 29 January 2006 22:33<br>To: Aisling O'Driscoll; <a href="mailto:serusers@iptel.org">serusers@lists.iptel.org</a><br>Subject: Re: [Serusers] SER RTP QoS Monitoring
<br><br>Hello -<br> Your method #1 (RTCP) is the best solution, it would seem. Method<br>#2 (INFO) is sort-of supported in some devices, like the Cisco 79xx<br>series with SIP software - they support insertion of some basic call
<br>element data into the BYE message. Method #3 is complex, as it would<br>require the SNMP collector to have full knowledge of all call states<br>on the devices that were queried (though possibly this is not more<br>complex than the RTCP collection process, either.) In any of these
<br>three cases, SER does not collect the call quality data - the<br>endpoints or media proxies are the only places that see this data.<br>If you run a mediaproxy in conjunction with SER, then it may be<br>possible to collect and report on that data, but there are drawbacks
<br>to a design of that nature.<br><br> For the long term, I would suggest taking a look at this draft,<br>which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to<br>be transmitted via SIP messages between media-handling entities and
<br>other reporting systems:<br><br><br><a href="http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-">http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-</a><br>08.txt<br><br> While we are still a while away from seeing vendor (or open source)
<br>implementations, this seems like a reasonable proposal. It allows<br>for SIP-based publication or subscription of call quality events by<br>various network elements, and it will permit arbitrary querying of<br>this data at any time interval (not just at the end of the call.)
<br><br>JT<br><br><br><br>>Hi,<br>><br>>I want to be able to extract QoS parameters such as jitter, delay and<br>>packet loss from VoIP calls made using my SER(both peer to peer for<br>>my public clients and also for clients where the voice will be routed
<br>>with mediaproxy). I would like to develop something as opposed to<br>>purchasing a software package.<br>><br>>>From searching the Internet the last few days I have seen that the<br>>following options seem to be available:
<br>><br>>1) RTCP<br>>RTCP stream will contain this information. The problem is that<br>>sniffers cannot clearly distinguish RTCP packets from other UDP<br>>packets unless something like an IP address or specific UDP port is
<br>>supplied. I had considered using TCPDUMP but then where do I position<br>>this in the network when voice is going peer to peer between client<br>>and I'm not sure how this can work to monitor all calls when I might
<br>>not be aware of the end user IP addresses.<br>><br>>2) SIP INFO extension<br>>This also seem to be an option but most off the shelf phones don't<br>>seem to support this and this would required modification of a SIP ua.
<br>><br>>3) SNMP<br>>I thought maybe SNMP might be an option but the SER snmp module no<br>>longer exists...<br>><br>>Does anyone have any comments on the above? Are the statements that I<br>>made correct or can anyone think of other ways to monitor the voice
<br>>QoS? I am trying to understand how commercial applications have<br>>accomplished this.<br>><br>>Many thanks in advance for advice.<br>>Aisling.<br>><br>>>---- Original Message ----<br>>>From:
<a href="mailto:clona@cyberhouse.no">clona@cyberhouse.no</a><br>>>To: <a href="mailto:ashling.odriscoll@cit.ie">ashling.odriscoll@cit.ie</a><br>>>Subject: Re: [Serusers] Status of SER SNMP module<br>>>Date: Sun, 29 Jan 2006 20:30:02 +0100
<br>>><br>>>><br>>>>Hi Aisling,<br>>>><br>>>>The SNMP module is dead as a killed turtle :(<br>>>><br>>>>-Atle<br>>>>* Aisling O'Driscoll <<a href="mailto:ashling.odriscoll@cit.ie">
ashling.odriscoll@cit.ie</a>> [060129 20:26]:<br>>>>> Hello,<br>>>>><br>>>>> I'm just wondering what the current status of the SER SNMP module<br>>>is?<br>>>>> Is it currently supported and is there somewhere I can find
<br>>>>> documentation on it?<br>>>>><br>>>>> Many thanks,<br>> >>> Aisling.<br>><br>>_______________________________________________<br>>Serusers mailing list<br>
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