Whirlpool

From some of the recent threads there seems to be a need to summarise the process of selecting the right VOIP codec.

(Apologies for those that have read something like this before...)

Start with the easy one:

Rule 1 - Bandwidth

64k or less - you must use G.729, G.723 or alternative low bandwidth codec (ie insufficient bandwidth for G.711). VOIP over dialup can work but use of a calling card may give better results... :-)
128k or more - any codec you like (as long as your provider has adequate upstream bandwidth)

Note that cable network bandwidth is shared and hence upstream congestion may limit available bandwidth. In this case better results may be achieved with low bandwidth codecs (this seems to mostly affect BigPond cable)

Use QoS where possible otherwise avoid other bandwidth intensive activities (eg file uploading/downloading) whilst VOIP call is in progress.

Rule 2 - Use the best codec you can!

By decreasing MOS (Mean Opinion Score):
G.711a PCM (Telstra PSTN/ISDN standard for national calls) bw=64k mos=4.1
G.729a CS-ACELP bw=8k mos=3.9
G.723.1 MP-MLQ bw=6.3k mos=3.9
G.726 ADPCM (used by all DECT cordless phones) bw=32k mos=3.85

NB Yes there are other codecs (eg GSM) but they are not widely supported/available in VOIP world (yet)

For more codecs see:
www.cisco.com/warp/publi...plexity.html#mos
www.ixiacom.com/library/...key=voip_quality

Rule 3 - Codec tuning for destination (ie transcoding issues)

Use of more than one compression codec can seriously impact call quality and should be avoided where possible.
[This results from tandem codec performance. Some comparative performance data can be seen in: www.mat.ucsb.edu/~ggroup...globecom2004.pdf]

The main call destinations where an alternate compression codec is already in use are mobile phones. GSM mobiles typically use the GSM (!) EFR codec (bw=12k mos=4.0) - which is an ACELP based codec.
(ie connection is something like User phone - VOIP codec - Internet - VSP - G.711 - Carrier - GSM - Mobile)
Thus in this case G.711 (specifically G.711a for Australia/Europe) should be used if bandwidth is available.

NB A cordless phone often introduces another codec. As per info in rule 2, DECT phones use G.726 ADPCM @ 32k. This is why a DECT to GSM call over VOIP G.729a or G.723 sounds so much worse than a G.711a connection.

For most landlines any good quality codec (eg G.711a, G.729a or G.723) is fine.

Note that with some ATAs (eg Sipura) you can select the codec on a per call basis:
*017111 - use G.711a
*01729 - use G.729a
(May need to change dialling plans to permit use of codes etc)

Rule 4 - G.711a or G.711u?

G.711a is used within Australian and Europe
G.711u is used within US

Select G.711u *only* if you are making lots of international calls to US
Why? Within Australia the telco interconnects are mostly ISDN - and G.711a PCM is used for all interconnect encoding.
Use of G.711u will result in a *small* amount of call quality degradation due to transcoding.

Hope this is useful.

dB

Interesting info regarding the mixing of codecs - something I had not thought of.

this should be a sticky.........

I didn't know anything about this and now my calls are clear.

Thanks

Any ideas what happens once it gets to a VSP?

I would assume if you use alaw to connect to the VSP that they would normally use some form of compression to send the data overseas or even within their network.

For what it's worth the ITU G.711 standard has a table at the back of the document to indicate which values need to be changed when converting from uLaw to aLaw. Most of them are left unchanged in the conversion process. You probably won't notice any difference if you use uLaw on your ATA/softphone for calls to a PSTN that uses aLaw (Australia) but it seems to be a pointless waste of time.

So, kids, in Australia use G.711a for the best call quality. If you are using G.711 to do faxing make sure you use aLaw!

Jim Bookis writes...

So, kids, in Australia use G.711a for the best call quality

When I connected with iiNet I got an email stating: "Preferred Codec Order: G729, G711ulaw, G711alaw "

I am wondering why they prefer G711u before G711a.......

I have configured my SPA3000 to use G711u but the emails above suggest that this is not optimal.

david writes...

Hope this is useful.

Yes, and a great effort it was too. However, you didn't have to completely re-invent the wheel, (edit) as much of this is covered already, in the Whirlpool VoIP Provider Wiki at the top of the page.

Cheers.

what's up skip writes...

Any ideas what happens once it gets to a VSP?

I would assume if you use alaw to connect to the VSP that they would normally use some form of compression to send the data overseas or even within their network.

It depends on the VSP. In the Internode forum they said they use alaw the whole way, even for overseas calls. So in my case it goes:
IP phone (alaw) --> Asterisk (alaw) --> NodePhone (alaw) --> PSTN (alaw???)
and doesnt get converted along the way.

If you find out what codec your VSP is using then you can use that codec all the way from your IP phone/ATA, and avoid the loss in quality as a result of transcoding.

FaktorTel - ULAW all through the Backhaul.

Chris.

U-238 writes...

If you find out what codec your VSP is using then you can use that codec all the way from your IP phone/ATA, and avoid the loss in quality as a result of transcoding.

I think that a field about that should be inserted into each page that describes a VSP in the Wiki. Here is a link to such an existing page so you can see what I am getting at: index.cfm?a=wiki&tag=VOIP_Faktortel

What do you think of this idea?

BTW, when one uses a dialplan with two or three VSP's, getting a single codec match for all of them right through to termination points could be a challenge.

Cheers

If using Faktortel....

What codec and protocol would you recommend..

Ussing iinet and getting 11.1m up and 1.02m down.

Thanks

Roden writes...

When I connected with iiNet I got an email stating: "Preferred Codec Order: G729, G711ulaw, G711alaw "

I am wondering why they prefer G711u before G711a.......

I have configured my SPA3000 to use G711u but the emails above suggest that this is not optimal.

I am taking a stab here, but most providers would use equipment manufactured by US companies, Cisco, Quintum etc. I would assume that this is the default codec preference.

FaktorTel do promote ULAW but when I just went to my account page to change my codec ALAW was there so I changed it.
I haven't tested it yet.

Cheers
Hally

AlexErko writes...

BTW, when one uses a dialplan with two or three VSP's, getting a single codec match for all of them right through to termination points could be a challenge.


I think you may be right there. I have FaktorTel as my primary provider and SPANtalk as my gateway 1 provider and Sipme as my gateway 2 provider.
I use FT for local calls and ST for national calls and SM for mobile calls, FT and ST both use G711 but SM don't...I made a call earlier using ST and I watched what was happening on my configuration page with my SPA 3000 and the call went through as G729....I was wondering about that.

Cheers
Hally

Jim Bookis writes...

For what it's worth the ITU G.711 standard has a table at the back of the document to indicate which values need to be changed when converting from uLaw to aLaw. Most of them are left unchanged in the conversion process. You probably won't notice any difference if you use uLaw on your ATA/softphone for calls to a PSTN that uses aLaw (Australia) but it seems to be a pointless waste of time.

Yes the differences are subtle (slightly higher dynamic range for A-law IIRC) but there seems to be a bit of anecdotal evidence on this forum and others that small quality improvements result from use of A-law with some VSPs.

Of course we're assuming that all equipment manufacturers have correctly implemented the two standards - I wouldn't be too surprised to find out that there is no difference in some low cost implementations... :-)

dB

AlexErko writes...

Yes, and a great effort it was too. However, you didn't have to completely re-invent the wheel, (edit) as much of this is covered already, in the Whirlpool VoIP Provider Wiki at the top of the page.

Yes most is already covered but I didn't see any part on tandem encoding issues.

No doubt you'll show me an example ... ;-)

dB

FaktorTel Guy writes...

FaktorTel - ULAW all through the Backhaul.

Why? Does uLaw get shoved onto the Australian PSTN ISDN gateway without conversion and viceversa?

david writes...

No doubt you'll show me an example ... ;-)

There are none, as that part of your post was the really tasty stuff that is not in the Wiki.

Cheers.

david writes...

Yes the differences are subtle (slightly higher dynamic range for A-law IIRC) but there seems to be a bit of anecdotal evidence on this forum and others that small quality improvements result from use of A-law with some VSPs.

I can't remember which but one of them maps to 14 bit linear and the other 13 bit. I think A-law is 14 bits. Mu-law does whacky bit flipping on alternative bits too. I'd love to know the 1960's engineering rationale for it. Probably improved the spectral spread on microwave radio trunk links.

I just made a call using ALAW and the call quality was horrible :-(
Back to ULAW...

Cheers
Hally

Hally writes...

I just made a call using ALAW and the call quality was horrible :-(
Back to ULAW..

Which provider are you using? If you make a call to the PSTN using A-law (G.711a) the audio should be excellent and totally unmodified all the way to the line card in the exchange at the far end. Something in the middle is wrong.

Not that something in the middle being wrong is unusual. If you have a CDMA phone and call another CDMA phone, the data is compressed (quality loss), uncompressed at the CDMA switch (more quality loss) and recompressed again (more quality loss) before going to the other to CDMA phone be uncompressed again (more quality loss)! I asked why the orginal CDMA compressed data didn't just get passed on to the other end without a de/compress cycle and got a mumbled reply about software upgrades and being too hard or something...mumble...

I don't know if this happens as well in GSM switches on GSM to GSM calls. I hope not!

I was using FaktorTel for that call and haven't had call quality issues with them at all lately and I monitored the call and it was definitely G.711a and there was a lot of echo and crackling and other party complained that echo was so bad she could barely understand me.
I just made a support call to SPANtalk and they told me they were receiving the call via G711a and the call quality was pretty good. I don't have them forced to any codec.
With FT I can enforce a codec that when I dial using them it will only use that codec.
I do this through my account page.
Cheers
Hally

Does any one know what other providers like ifone, Oztell & MyNetFone do?

See www.cisco.com/en/US/tech...00801149b3.shtml

with regards to G711a vs G711u.

OzTell supports iLBC, Ecomtel as well.

We are trying to get ZyXEL to support iLBC, and Dynalink later. No guarantee, but we will try, cos iLBC is better

Chicken writes...

cos iLBC is better

yes I find iLBC is less affected by upstream transcoding than G729.

one thing i've noticed, which didn't occur to me before but is of course logical, changing up to a higher quality codec won't improve a noisy service, the noise level will remain the same as i've found with my Faktortel line

i've tried all of them, from the lowest to the highest codec

i've no idea what codec Spantalk use (can't see anywhere on my account page to change, not that i'd want to) but on my 512/128 connection it's always been perfect sound, clean and crisp, even with email checking in the background and with slow BT transfers running also

i don't have any QoS control at all

dddp writes...

i've no idea what codec Spantalk use

well i do now, i just made 2 calls, 1 via spantalk and 1 via faktortel, my PAP2 showed the spantalk call as G729A and showed the faktortel call as G711a (i actually have ALAW selected in "my account")

that's interesting, i thought spantalk was G711a, great quality at that codec setting

This thread is starting to confirm what I have experienced for some time. That is that (edit) when a lousy connection &/or a lousy termination leg &/or other non-codec issue reduces the quality of a call using a G.711 codec, that call can sound much worse than a call using a very compressed codec such as a G.729 or even a G.723.1 where those lousy elements don't exist.

In fact when all those non-codec issues are not present my subjective experience has been that there is very little difference in the quality of a call whatever commonly used codec is used.

Cheers.

dddp writes...

my PAP2 showed the spantalk call as G729A and showed the faktortel call as G711a (i actually have ALAW selected in "my account")

I have been monitoring calls also...all Spantalk calls go through as G729a and I changed FaktorTel back to G711u and sound quality improved dramatically. The ALAW didn't work well for me with FaktorTel. My guess is it is the equipment they are using.....just a guess.

Cheers
Hally

Good post.

Just a little codec info for all

G.711 ITU standard for ISDN 64 kbps
G.723.1 Pass through mode only 6 kbps
G.726 Only 32kbps Version is supported in asterisk 16/24/32/40 kbps
G.729 License required 8 kbps
GSM Mobile telephone codec 13 kbps
ILBC Internet low bandwidth codec 14 kbps
Speex Configurable codec 4-48 kbps
LPC10 Linear predictable codec ‘robot’ 2.5 kbps

david writes...

G.711a PCM (Telstra PSTN/ISDN standard for national calls)

Does this mean that if I made a call from an IP phone using 711a, to a VSP using 711a, which then passed the call down an ISDN line (which you say uses 711a), the call would not be transcoded at all throughout the entire path? Not even decompressed and re-encoded at the VoIP-PSTN gateway?

Does this mean that if I made a call from an IP phone using 711a, to a VSP using 711a, which then passed the call down an ISDN line (which you say uses 711a), the call would not be transcoded at all throughout the entire path? Not even decompressed and re-encoded at the VoIP-PSTN gateway?

That's right. It (G.711a) should pass unfettered from your ATA to the line card at the other end's exchange. These VOIP monkeys who insist on you using G.711u in preference to G.711a clearly buy their backend gear from the USA and don't bother changing it or even finding out what ACA requirements are. They obviously work on the "It sounds OK" principle.

Jim Bookis writes...

These VOIP monkeys who insist on you using G.711u in preference to G.711a clearly buy their backend gear from the USA and don't bother changing it or even finding out what ACA requirements are

hmmmmn.....it gets curious and curiouser....

Hally writes...

My guess is it is the equipment they are using.....just a guess.

Faktortel Guy clearly said ULAW all the way through backhaul. Wouldn't that mean he is suggesting for us to use ULAW where possible?

I have been using G711 (random between A and U - tell no difference between them) with X-Lite. But now with the PAP2 and using a standard(ish) touchphone it seems better quality with G729. It could just be Voise, though. (ref forum-replies.cfm?t=410937&p=2#r21)

Maybe I need to try G711 with the PAP2 and see what it sounds like...

Giant writes...

Faktortel Guy clearly said ULAW all the way through backhaul. Wouldn't that mean he is suggesting for us to use ULAW where possible?

He did and clearly for reason...when I tried ALAW it was crappy quality...when I reverted back to ULAW quality improved greatly.

Cheers
Hally

Thanks Hally,

Just wasn't sure why you tried ALAW. I guess we just need to experiment to get the best quality codec for our own unique situations. So many factors make a difference.

The "op" David's advice of trying to keep down the number of codec changes for any particular type of call, seems to make sense to me. I use Oztell and an SPA3k, and have set my Line1 and PSTN codecs to ALAW. In my Oztell setup I force the use of ALAW. Pity I use a dect phone on the SPA3K to muck up a good link. (it's not bad really) Might plug in an old touchphone to measure the difference.

Cheers
Giant

david writes...

Use QoS where possible otherwise avoid other bandwidth intensive activities (eg file uploading/downloading) whilst VOIP call is in progress.

Rule 2 - Use the best codec you can!

just a thought- when using a softphone and you want to use the larger codec 711u needing max bandwith. If the qos is disabled in windows (limiting reservable bandwith to 0) wouldnt this give an increased bandwith to the call by 20% I have done this and have crystal clear calls on softphone but but like david said if you do anything else it interferes with the call any ideas on this? ienabled qos in router.

that was interesting, just made a call with G729a spantalk, i had 2 P2P programs running also, 1 was BT which was monitoring something like 7 files, uploading one at 4kBs, downloading with the other P2P at 3-5kBs, surfing ebay as well and had no call quality problems at all at either end and no QoS control either

dddp writes...

that was interesting, just made a call with G729a spantalk

All calls that I have monitored with SPANtalk have been G729a and quality pretty good.
I even rang them and asked about the codec and they seemed a bit mystified as to why it was going through on that codec. They don't have the means to force calls through on any particular codec, you need to set your ATA to that and I have G711u as my preferred codec but not forced. I am not sure if that really makes any difference at all though.

Cheers
Hally

dddp writes...

that was interesting, just made a call with G729a spantalk,

Please tell us what the available bandwidth is on your ISP plan.

Cheers.

Hally writes...

All calls that I have monitored with SPANtalk have been G729a and quality pretty good.

All of mine have been G711a, and that's using X-lite, with nothing forced...

Edit: G711u, in X-lite, was placed in preference to G711a, so for some reason it's choosing a over u. And I don't have G719 to use in X-lite.

512/128 plan iinet

X Lite do tend to use different codecs to ATA's I have found.

Cheers
Hally

Giant writes...

Pity I use a dect phone on the SPA3K to muck up a good link. (it's not bad really) Might plug in an old touchphone to measure the difference.

The DECT issue only really matters when you have to use a low bandwidth codec (G.729a or G.723) to a non G.711 destination. Since you're using G.711a then degradation from a DECT phone will be minimal!

In fact a VOIP setup of DECT (analogue) - ATA - G.711a - VSP - destination should be superior to conventional DECT (analogue) - PSTN - destination ... simply because you've shortened the analogue component to metres instead of kilometres.

This is at least true for high quality VSPs who don't mangle G.711a.

dB

sparky writes...

If the qos is disabled in windows (limiting reservable bandwith to 0) wouldnt this give an increased bandwith to the call by 20% I have done this and have crystal clear calls on softphone but but like david said if you do anything else it interferes with the call any ideas on this? ienabled qos in router.

Unfortunately the way QoS is implemented in routers varies somewhat. The most common methods:
(i) switch port priority (all packets from a given port are set at a given priority)
(ii) application priority (ability to prioritise packets by application) - although the problem here is that some routers don't seem to know about RTP - and many routers lack the flexibility required to properly define a new application
(iii) TOS bit support - typically available only on more expensive routers

Best to check with the vendor of your router if it supports QoS for VOIP RTP.

dB

Hally writes...

They don't have the means to force calls through on any particular codec, you need to set your ATA to that and I have G711u as my preferred codec but not forced

If Spantalk have g729 as perferred and you have g711u as perferred, guess what you will probably get g729. You must disable the other codecs to force it to use your perferred g711u, this has the side effect of limiting your compatibility with other sip clients.

Newbei writes...

If Spantalk have g729 as perferred and you have g711u as perferred, guess what you will probably get g729. You must disable the other codecs to force it to use your perferred g711u, this has the side effect of limiting your compatibility with other sip clients.

Spantalk tell me that their preferred codec is G711 not G729 and Sipme use G729 so I am not about to shut down codecs for now.

Cheers
Hally

dddp writes...

that was interesting, just made a call with G729a spantalk

Do you get really clear calls with that codec? i found it a little choppy with iiNet so i forced all to g711u with AstraTel and iiNet and found that really clear and haven't tried other codecs yet but from what the other guys have said it might be worth the try

Lots of talk about codecs, but none about packet sizes. Yet altering the packet size can make a lot more difference to the call quality than changing codecs alone, depending on other factors such as latency and in particular big variations in latency during a call as is common with broadband wireless connections ......

Cheers.

sparky writes...

Do you get really clear calls with that codec?

yes perfectly clear, but only with spantalk, faktortel with G729a is fairly ordinary

how do you see what codec you are using when calling someone?
Using the Belkin Voip ...

Thanks

also Faktortel as voip provider

AlexErko writes...

Lots of talk about codecs, but none about packet sizes

has anyone looked at say a spa3000 which by and large seems to have US settings, and are they able to be configured any better for Australia or does it not matter.
in ths Spa3000 Sip Tab under RTP Paramaters (about halfway down the tab page) the size is listed as 0.030 - leave it as is or has anyone founda better size?

sounds also like a loose group meeting of all voip providers and equipment suppliers occasionally would be a good idea - to talk about overall improvement of customer acceptance and ease of setup to maximise the overall voip experience for the beginners and to minimise the overall frustration of the newcomers to Voip -only a suggestion not a criticism.

Also like to thanks those suppliers that visit this forum and provide the help that they do!

Cheers
Marty

bluebay writes...

has anyone looked at say a spa3000 which by and large seems to have US settings, and are they able to be configured any better for Australia or does it not matter.
in ths Spa3000 Sip Tab under RTP Paramaters (about halfway down the tab page) the size is listed as 0.030 - leave it as is or has anyone founda better size?
The key settings relating to Australia are areas such as line impedance and ring tones which are well documented elsewhere.

In relation to packet size, AlexErko is right that this parameter can improve quality in some situations.

The key network issue that this parameter can help with is packet loss. When packet loss occurs, most compressing codecs will first attempt to hide the missing data (interpolation) and then beyond a certain point an audible dropout/click is heard. A good example is the GSM EFR codec which hides temporary radio signal loss quite well but eventually the point is reached where the speech becomes unintelligible.

With networks having a small *constant* rate of packet loss (1% or less) reducing the packet size will increase the probability that an encoded speech sample will make it through to the receiver. This increases quality is situations where packet loss is occurring.

However with some providers any increase in packet rate may also result in a proportional increase in packet loss... This is particularly the case with contention based network technologies such as cable (ie DOCSIS).

So, as with any tuning approach, change the packet size in small steps. Currently I'm using 0.02 on my SPA-3000 with OptusNet cable and it works okay. Note that some VSPs fix the packet size on the return path.

sounds also like a loose group meeting of all voip providers and equipment suppliers occasionally would be a good idea - to talk about overall improvement of customer acceptance and ease of setup to maximise the overall voip experience for the beginners and to minimise the overall frustration of the newcomers to Voip -only a suggestion not a criticism.

As I understand it ACIF has been working with VSPs to develop a self regulatory approach. However given the current poor state of affairs with some providers unashamedly ignoring existing Aust technical standards there is a pressing need to sort this out before the wider net community enters the VOIP market.

dB

OK, so does anyone have a great idea on how to select a particular codec on an outbound call from Line 1 with a SPA-3k? The default is G711u, which by the OP's standards is inferior.

Edit: and the obvious solution, dialling *027111 before the number, does not work on 3.17 or 2.13.

david writes...

So, as with any tuning approach, change the packet size in small steps. Currently I'm using 0.02 on my SPA-3000 with OptusNet cable and it works okay. Note that some VSPs fix the packet size on the return path

OK , so I guess you probably changed from 0.03 to 0.02 and noticed a change in quality. I think I will try it and see.

As I understand it ACIF has been working with VSPs to develop a self regulatory approach. However given the current poor state of affairs with some providers unashamedly ignoring existing Aust technical standards there is a pressing need to sort this out before the wider net community enters the VOIP market.

I think self -regulation would be a good step for Voip service providers overthe next 6 -12 months - better to be seen to sort thing out a little as an industry than wait for a government list of compliance issues.

Thanks David, most helpful !

Cheers / Marty

OK, so does anyone have a great idea on how to select a particular codec on an outbound call from Line 1 with a SPA-3k? The default is G711u, which by the OP's standards is inferior.

I opened a ticket with Sipura/Linksys/Cisco support on this - there is currently no way to automagically select the right codec via the dial plan. Hopefully they may add this feature one day...

Edit: and the obvious solution, dialling *027111 before the number, does not work on 3.17 or 2.13.

This works for me on 3.1.7(GWg). What is your current dial plan?

dB

I have amended point 3 in the VoIP Quality check-list Wiki to include a reference to transcoding / tandem encoding and a link to this thread. See index.cfm?a=wiki&tag=VOIP_Quality

Cheers.

I have tightened up the wording in Point 3 of the VoIP Quality Wiki page.

Cheers.

Jack O'Neill writes...

how do you see what codec you are using when calling someone?
Using the Belkin Voip ...

not sure bout the belkin, but with a PAP2 its the on the very 1st page of the configuration page under line 1 status, call a number then during the call refresh this page, it shows quite a lot of info, sorry you had to wait 10days for someone to answer you

whoozle writes...

G711u, in X-lite, was placed in preference to G711a, so for some reason it's choosing a over u.

Same thing happened to me. Am using X-Lite coupled with Pennytel and Spantalk. Disabled all other codecs apart from u which with Pennytel is just great. Have been forcing Spantalk to use u and that seems to have been a bad move. Problems with noise/issues etc. Only forced u as that is where the best comments came from my friend testers in regards to call quality.

Just tested it by enabling both g711u and a and letting the provider choose. Pennytel still defaults to u while Spantalk chooses a. Suspect that it is the equipment of the VSP's. Spantalk seem to be more Australian centred if one can put it that way where as Pennytel are a bit more international. Badly worded but you get what I mean.

sparky writes...

so i forced all to g711u with AstraTel and iiNet and found that really clear

Has ii confirmed their preferred codec?

I find that Astratel tends to implement g729 with my calls. The quality isn't really an issue.

Westsurf writes...

I find that Astratel tends to implement g729 with my calls. The quality isn't really an issue.

Hi Westsurf,

The details below might be helpful. I'd suggest using G.711a over G.711μ since the terminating gateways connected to the Australian PSTN all use G.711a.

The codec’s supported by Astratel and their selection order are listed below.
• G.729 Annex A
• G.723.1 – high and low
• G.711 a-law
• G.711 μ-law

Just select your preferred codec to the top of your ATA's codec selection order.

We have G.729 and G.723 as the higher preferenced codecs since they cope better with the inevitable packet loss over the internet than G.711a/μ.

AstraTEL writes...

We have G.729 and G.723 as the higher preferenced codecs since they cope better with the inevitable packet loss over the internet than G.711a/μ.

Gday Adam,

Currently, I have iiNet as my main VSP (local n mobiles). However, the untimed Australian calls with Astratel grabbed me, so Astratel gets all the non-local calls.

I haven't tried the g.711a yet. As most calls go via G.729 with Astratel and my iiNet calls go via g.711u - I haven't realy had a quality issue with either.

I do get crap calls occasionally, at .10 a call, hang up try again.

I will give the G.711a a go when I have more time (not Xmas/New Year) to work out the best 'preferred' codec at my end.

Thank you for the response.

westsurf

bluebay writes...

in ths Spa3000 Sip Tab under RTP Paramaters (about halfway down the tab page) the size is listed as 0.030 - leave it as is or has anyone founda better size?

Changing the packet size from 0.030 to 0.020 has made a huge difference in quality with g711 codec on Astratel. Perfect sound quality on both ends whereas at 0.03, receiver complained of static and my voice breaking up.

NEXIUM writes...

Changing the packet size from 0.030 to 0.020 has made a huge difference in quality with g711 codec on Astratel

Would this change make any diference (better or worse) using the G729 codec?

Giant writes...

Would this change make any diference (better or worse) using the G729 codec?

I don't think so as G729a is compressed format so probably does not need such a small packet size to maintain reasonable quality. I could use g729 at 0.03 with no problems. Only g711 did not work very well.

Thanks Nexuim, I'll leave it as it is then. Working quite well, but I'm always looking for a "go-better" tweak.

Giant writes...

Would this change make any diference (better or worse) using the G729 codec?

Hi Giant,

Probably not a whole lot. From past experience, changing the packet size can help give you better results but is a bit of trial and error.

You might be able to change the packet size mid call so making it easier to check if the change helped to make it better or worse. It depends if you ATA can support mid call packet size changes - The Drayteks deal with it fine but I think the Sipuras *might* save the settings and reboot or re-register stopping you from easily seeing if the change is good.

We find for lower speed lines that a larger packet size may help. Also for customers further away from us seem to benefit from a larger packet size.

If the packet size is too high then you may start to hear more of a two-way CB type call when there is some noticeable delay. I'd suggest in starting on a lower value and and a listen to the call. Wind up the values until it sounds acceptable. Try it at the maximum value just to compare. G711, G723 and G729 allow the packet sizes to the altered and shouldn't cause problems with other ATA setttings.

If your ATA makes calls to several providers (who probably have gateways spread far and wide and not nearby) then setting the packet size at 'x' ms might sound good for one but not for the next. The packet size on most ATAs stay the same regardless of which provider is taking your calls.

It is an individual tuning thing. What works for one may not work as well for the other. Most ATAs are within 20ms to 40ms. Some are higher but as along as it sounds good that is the main point overall.

Hope this helps,

david writes...

The DECT issue only really matters when you have to use a low bandwidth codec (G.729a or G.723) to a non G.711 destination. Since you're using G.711a then degradation from a DECT phone will be minimal!

I am a little confused about the above statement... I am with MyNetPhone and they only support G.729a, but i use a DECT cordless phone. and i have noticed that the quality is not that good.

First question... Do i need to change provider to someone who supports G.726 or G.711?

Second question... If i do change provider to someone who supports both... Should i use G.711 or G.726?

Thanks in advance if anyone can help... i am a little mystified...

AEKARA_OMDARA writes...

I am with MyNetPhone and they only support G.729a, but i use a DECT cordless phone. and i have noticed that the quality is not that good.

Hi AEKARA_OMDARA,

I think the DECT phones also use voice compression (G726 ADPCM), so your speech probably:
* compresses with G726 within your DECT handset
* decompresses back from G726 when it leaves the DECT basestation
* compresses again with G729 when it reaches you ATA/VoIP modem
* decompresses again from G729 when it reaches the telephone network.

There might be a further compression hops depending where the call is sent over the PSTN. Both of these codecs are lossy but do save some bandwidth.

First question... Do i need to change provider to someone who supports G.726 or G.711?

No, I'd try a few calls using a plain ye olde corded standard telephone before anything else. The DECT phones aren't a bad thing, but coupled with VoIP where both compress and have some added delays etc you can loose quality. YMMV of course.

Roden writes...

I am wondering why they prefer G711u before G711a....

Most ATA's are factory defaulted to the USA G&711u codec the ,g711a codec is used for voice in Australia and often the g711u codec for Fax.

Thanks AstraTel..

I have tried a plain corded phone, and a non-dect analogue phone... and the quality is better, but it isn't that noticeable... i really want to use the DECT phones as i use a nice set of 3 dect phones, which are handy for around the house.

I am using optusnet cable, and bandwidth isn't a problem so i would like to know, if the quality of using DECT over a provider with G711 or G726 would be better than using a DECT phone over a provider who only supports G729.

BTW.. the problem seems to be more on the non-voip side... i.e. i can hear the other party fine, but the PSTN party can't hear me very well... the volume is low and a little croaky...

Any further ideas.?

Cheers,
Paul

AstraTEL,

Hi, I am having the same problem with MyNetFone & DECT Cordless.
Will be interested to know if there is a solution, I too don't want to go back to a standard phone,

Pinders2000 writes...

AstraTEL,

Hi, I am having the same problem with MyNetFone & DECT Cordless.
Will be interested to know if there is a solution, I too don't want to go back to a standard phone,

Good luck getting any answer from AstraTEL.....

forum-replies.cfm?t=528658

Interesting about DECT though. I had no idea that my DECT phone was a possible weak-link in the chain.

I use G711a via a $40 Panasonic DECT cordless handset and have had no call quality issues, but I've never actually tried it with a non-cordless phone either.
Maybe I'm missing out. Have to dig that old Telstra land-phone out of the cupboard and give it a whirl.
But ATEOTD, you can't beat the convienience of being able to walk-about anywhere and talk to family interstate for 3&1/2 hours for 10c :-D

Ahhhh....AEKARA_OMDARA