Implementing Decent QOS for VOIP

air

Ok,
My setup is as follows:
Eircom DSL 128/2048

I originally got a Linksys RT31P2 as it purportedly did QOS but when I got it I found that the QOS was pretty poor on it and recently purchased a Linksys WRT54G in order to use it as a proper router.

Firstly my 128kb/s upload is actually 56kb/s by my calculations.
The max FTP upload speed I can achive is 7.0KB/s which in my book is:
(7x1024x8=57344bits /s = 56Kbits/s)
So can I complain to eircom for giving me less than half what I'm promised?

Ok now for the QOS settings, I'm using the Freeman / Talisman firmware and on first impressions I think it does do the QOS properly (it uses IPtables I think).



This was my setup originally when I was allowing roughly 80% of my upstream bandwidth as apparently I'm supposed to.

When doing this and uploading a file I was getting the full 7KB/sec speed, when I made a call (G7.11u 64K bandwidth) to the blueface echo test (301) the quality of the call was pretty ok, but the bandwidth available to the upload was more or less cut off. This wouldnt be particularly acceptable to users here who would want to be sending emails etc.

Once the penny dropped regarding my actual bandwidth I reduced the setting in the QOS config of the WRT54G to 40Kb/s upload and tried again.
This time I was using G726.24 (24Kb/s) codec.

Now on sustained (reasonably large) uploads the speed is capped at about 4.5KB/s which at least leads me to believe that the packet filtering is working and the router is restricting my upload to what I've told it is available.
This time the jitter on the call was a bit unacceptable (still no real delay) and the download got cut off untill I hung up when it returned to normal.

As I see it, I have a number of problems:
1. Eircom are not delivering their specified amount of upstream bandwidth
2. Their UL/DL ratio is ridiculous
3. The various voice codecs are either using more than their specified bandwidth or else the associated filtering is causing an amount of packet loss which is an unacceptably high proportion of the remaining bandwith available when provision has been allowed for between voice & data traffic.

Any feedback, tips, corrections or advice welcome

causal

air wrote:

Firstly my 128kb/s upload is actually 56kb/s by my calculations.
The max FTP upload speed I can achive is 7.0KB/s which in my book is:
(7x1024x8=57344bits /s = 56Kbits/s)
So can I complain to eircom for giving me less than half what I'm promised?

If it's that bad - dialup speed - then hell yes complain.
Check your connection speed here
Also check your ping times to boards.ie and jolt.co.uk
- post all of the above results - it's only a snapshot but it might help

As I see it, I have a number of problems:
1. Eircom are not delivering their specified amount of upstream bandwidth
2. Their UL/DL ratio is ridiculous
3. The various voice codecs are either using more than their specified bandwidth or else the associated filtering is causing an amount of packet loss which is an unacceptably high proportion of the remaining bandwith available when provision has been allowed for between voice & data traffic.

Any feedback, tips, corrections or advice welcome

1) Apparently so - but let's see those test results.
2) Yes - but that may change IF this thread is true. But I wouldn't hold my breath.
3) I doubt the codecs themselves usem more than specified. However, I can't remember if VoIP packets send acknowledgements (like TCP does) or if they're sent as datagrams (fire and forget) - if the former then there will be a slight overhead as you send acknowledgements. Add to that the SIP messages going back and forth. So your bandwidth usage will in practice always be higher than the codec itself.

hth,
causal

aaronc

air wrote:

3. The various voice codecs are either using more than their specified bandwidth or else the associated filtering is causing an amount of packet loss which is an unacceptably high proportion of the remaining bandwith available when provision has been allowed for between voice & data traffic.

Any feedback, tips, corrections or advice welcome

The 24K bandwidth quoted on the Linksys RT31P2 only refers to the bandwidth required for the payload.

If you ignore Ethernet headers assuming that you don't have a bottleneck on your private LAN (this may not always be the case especially if you're using WiFi) and look only at IP you have:

IP Header: Typically 20 bytes (RFC 791)
UDP Header: 8 bytes (RFC 768)
RTP Header: Typically 12 bytes (RFC 3550)
Payload: 60 bytes*
Total Per Packet = 20 + 8 + 12 + 60 = 100 bytes / packet
Bandwidth = 50* x 100 = 5000 bytes/s = 40000 bps = 39.06 Kbps**

*G726 24Kbps takes 15 byte samples every 5ms and has a payload size of 20ms resulting in 50 packets per second with each packet having 60 Bytes.

**And I may have missed something here as well as the two references below give the rate as 47.2 kbps. It's possible that they're including the Ethernet headers as well.

http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml

That may go way someway towards explaining why your using more bandwidth then you thought. When you're sending a lot of small packets the IP/UDP/RTP overhead can start to have a significant impact.

I would also recommend you try G729 it uses even less bandwidth then G726 and has a reputation as being a very good low bandwidth codec.

Aaron

air

Will check it out on monday.
Incidentally, a 128/1024 home package here it's reporting 100/834.
With regard to the signalling overhead etc, I was thinking more along the lines that my 4 different filters (ip, mac, service, port) might be a bit over agressive.

On another point, I'm starting to think that the QOS on the RT31P2 might be better than I thought, I might try it again with revised upstream/downstream settings based on the results of my speed test.

Also I'm not using the Wifi at all, I have it disabled on the box afaik & have the antennas removed for good measure

The increase in upstream bandwidth would be a massive help to VOIP imho, the current situation on my own line is almost back to the early days where you couldnt use the phone & net at the same time!

air

Ping to boards - 27ms
Ping to Jolt - 37ms

Speedtest:

With Packet Filtering (QOS) Enabled:
Download: 1.52Mb/s
Upload: 35.8Kb/s

Without Packet Filtering:
Download: 1.71Mb/s
Upload: 100Kb/s


However I would reckon that the speed test doesnt test the upload long enough to get a true figure. A sustained ftp upload straight after the speed test managed exactly 7.0Kb/s as usual.

causal

Those ping times are fine.
The Packet Filtering certainly seems to be thottling your down/upload speeds; the question remains whether or not it uses the 'other part' of your bandwidth for you VoIP calls.

If your upload is 7KB/s (56kb/s) and you're only getting 100kb/s (rather than 128kb/s upload from your ISP) that leaves you with 44kb/s for your VoIP call - Aarons figures show that you need between 39-47 kb/s.
If all of the above is accurate then you're right on the limit - maybe try increase the bandwidth available for VoIP (or get onto Eircom) or as Aaron suggested try a different codec (G729).

causal

capistrano

Air, I also use a WRT54G router and I recently upgraded the firmware to v4.00.7, which has improved QoS compared to earlier versions.

And from my own experience I have to say that the QoS is much better. I was recently took a call while doing a big BitTorrent download and I could see the BitTorrent upload/download speeds fall pretty quickly and the call quality was fine. With earlier versions I could never do both successfully at the same time.

air

Sorry lads, had a bit of a brain fart, the upload speed was capped within my ftp client so that was restricting the bandwidth further than it should have been.
I reckon I probably do have about 100Kb/s alright, I've set the upstream to 90 in the QOS settings and seem to be getting good quality during uploads etc now.

I've even been using voipbuster today & the quality is perfect, although I dont know how long that would last .

I hope to start using the blueface line regularly and once I'm happy with it for a decent length of time, i'll hopefully transfer one landline number to it.