VOIP Tech Chat → Hi-def audio through VoIP.ms?

Hehehe I second that! Nice word, gotta remember it.

I love how people think that spewing fire at a company is going to make them run and please them. It'll just make them (As well as everyone around you) want to drop you like a hot potato.

Could you elaborate as to why?

It's a little bit more complicated than that. You see, the provider has two legs in a VoIP call in that situation. It's not peer to peer.

The provider maintains a voice leg with you, and then connect the call via SIP URI to the other provider.

Allowing G722 and not rejecting calls would lead to the following situations

1 - Transcoding down to another codec for the other leg of the call when it's not supported by your destination.

2 - Transcoding of all calls if you have selected G722 only on your device.

3 - Transcoding and extra bandwidth leads to higher CPU use and cost, making it impossible to maintain our low rates.

It has nothing to do with our competence

By the way, you can call the URI of your choice directly from your softphone, and you won't have to put VoIP.ms in the middle.

format for X-lite:
string@host

format for Zoiper:
sip:host/string

For other devices, you can consult your manual

Earlier in this thread there was the assertion that "you don't need a provider for peer to peer SIP calling." I don't think that this notion gets quite enough play or understanding.

SIP does not rely upon service providers in the same fashion that we relied upon telcos. You can place SIP calls directly between SIP end points.

However, this is often inconvenient because of discovery and network security issues. Thus it can be very useful to use an out-of-house SIP registrar, of which there are many...some completely free.

As it happens I use OnSIP as my primary SIP Registrar, but I have experience using gizmo5, FWD (now defunct), IdeaSIP, SIPGate, and SIPSorcery.

While it may be possible to use some of the more traditional ITSPs as SIP registrars, if HDVoice is not specifically in their product strategy you may find that they proxy the media and so force calls into G.711 when it's not absolutely required.

Since every G.722 capable IP phone I'm aware of is capable of at least two SIP accounts there should be not trouble registering with both a traditional ITSP and a free SIP registrar supporting G.722/video/whatever.

There also seems to be some idea that you need an Asterisk or Freeswitch instance locally to use peer to peer SIP calling. This is completely untrue. Such things are possible, but not required.

Using a local PBX adds a great deal of flexibility in routing calls once inside your LAN, but at the cost of adding considerable complexity as well.

My own experience has the knowledge overhead of admin tasks on a local Asterisk box were considerable. While it was nice to have the flexibility of a local dialplan, it gave me lots of room to make mistakes.

At the time, the overall reliability of my installation was defined by my relationship with Asterisk. In the end I found that using a suitably flexible SIP hosting service was more reliable than my own PBX.

That said, Barrett Lyon long ago did a great post about setting up SIP calling peer-to-peer using Asterisk:

»www.blyon.com/blog/index ··· dialing/

You may also find my posts on Making Use of HDVoice Right Now! relevant, as this invariably involves peer-to-peer calling.

»www.mgraves.org/2009/06/ ··· ght-now/

CallCentric can be useful for many customers in this area as well. I believe that CallCentric supports wideband calling in three ways:

1) For calls from one Callcentric user to another CC user, any codec can be used, including wideband (G.722). This includes users of the CallCentric "IP Freedom" accounts, which are totally free of charge.

2) The same applies to SIP to SIP calls made by a CC customer. Thus a CC user can make a wideband SIP URI call to someone who is not using CC. (Of course the guy at the other end has to be capable of receiving it!)

3) This would also apply to iNum calls, as the iNum network generally supports wideband (G.722) calling, depending on providers. It definitely applies to iNum calls between two CallCentric customers.

Wideband calling does require a lot more bandwidth. It could present a challenge when dealing with outside data centers, both in terms of expense and in terms of rapid fluctuations in bandwidth availability. This may be why it is not supported by all providers. CallCentric, operating its own facilities and having direct peering with most major backbone ISPs, would seem to have an advantage in this area.

PBXes also supports G722.

We've become such a society of extremes.

Half of the folk are quite content with the awful voice experience of cellphones.

The other half want to go to wideband, where you hear in painful crystal clarity the voice of someone you often would rather not be talking to in the first place!

--------------------------------------------

No middle ground. No happy medium.

PertPlus used to make their shampoo in a "Happy Medium" but they've dropped that name. Actually, P&G ditched the whole Pert brand and it's now owned by some Texans. The product is just not the same.

Mysterious. Just as strange as when Colgate abandoned the Halo Shampoo product, one of the world's best known brand names in the 1950-1960's.

Well, bad shampoo may leave you feeling limp, but CallCentric never will.

So how do I get the SIP client to make a P2P SIP call? I'm not sure that many SIP clients even support this. I use Bria on the PC and iPhone, and CSIPsimple on Android. I have a Yealink T26P IP phone on order, don't know yet about it's P2P ability.

The SIP clients all appear to want to register with a domain, username, and password. Bria won't even let me type in a SIP URI without registering to a server.

I am trying to figure out how to proceed to be able to dial the usual G711u PSTN calls, paying a provider for termination, who can port my old telephone number, while still retaining the ability to all a SIP URI using G722 or Speex wideband. Bonus WAF points if this doesn't require reconfiguring the phone between call types.

I think I understand that I want to connect my domain name to an SRV record pointing to my dyndns.org URL to staticize my dynamic IP. I would then use SIP URI me@mydomain.com.

However, my smartphones are not always used from mydomain.com. For example, I could be using wifi at work. What would I need to do to receive calls on the smartphones away from home? Register with Callcentric or OnSIP? No way to make my SIP URI work when the smartphone is away from home?

Do companies like Callcentric and OnSIP 'proxy' calls? Not sure if this is the right terminology, but it appears that voip.ms is doing something with the data stream that requires them to be able to decode it. Is the voice data actually relayed through their servers, rather than going directly between my SIP client and the recipient?

Lastly, I want to disagree with the hi-def codec BW useage is excessive comment. G722 uses the same as G711, and Speex wideband uses less. They do use some CPU, but who doesn't have enough CPU these days?

I love the new 24-bit sound formats on blu-ray. I don't buy MP3's, I still buy the CD. The cellphone/MP3 crowd can keep their horrible sound.

In the case of G.722, it typically requires about the same digital signal processing (DSP) horsepower as the narrowband ADPCM G.726. However, the trade-off with wideband G.722 is that it requires a data rate of 48, 56, or 64 kbps due to the larger sample rate of the voice channel. In comparison, narrowband codecs like G.711 can operate at data rates of 32 kbps and lower.

»www.eetimes.com/design/s ··· P-codecs

G.722 while the most common is also the most elderly of wideband codecs. It's worth noting that from AMR-WB to CELT, SILK, G.719, et al the entire universe of wideband codecs operate at equivalent or lower bitrates than the legacy PSTN standards. Only those that support full-band or stereo coding go beyond that 64 kbps of G.711.

The use of wideband is not so much a bandwidth issue as it is as transitional issue. That is, when a wideband capable device tries to call the PSTN either the device gets negotiated down to a narrowband codec or some gateway in the path performs transcoding.

Michael

I've been comparing OnSIP and Callcentric. OnSIP looks extremely capable, but at $40+/month it seems overkill for two numbers for home use. Running my own PBX might cost less, but I'm not sure if I want to bother with that just to get wideband voice.

I'll see if I can get Callcentric to work successfully with wideband. It may be the biggest issue is to get the SIP clients/phones to cooperate and call P2P SIP URI.

SIP URI syntax for Bria is address@host. read your manual.

I did not ask what SIP URI syntax is. Read the question.

Edit: Hint - "P2P"

You have such a nice way with people!

You already strongly implied that MartinM's company was not competent, in an earlier posting.
»Re: Hi-def audio through VoIP.ms?

Now your latest response reeks of sarcasm, condescension, and contempt.

I'm sure that everyone will rush to help you now.

No sir. That is simply not true.

The worst case wideband call would be using the ancient G.722 codec, which has exactly the same bandwidth requirements as the standard G.711 codec upon which the bulk of the PSTN resides.

The very fact that all of the wideband codecs are built using more modern approaches to compression means that they squeeze more audio channel out of a given data rate.

As this assertion keeps coming up again and again let me be 100% clear:

With respect to voice calling, the entire universe of wideband codecs at worst use the same amount of data bandwidth as the legacy PSTN codecs that they replace. Often the actually use less bandwidth.

Want the numbers? Look here for a summary:

»www.mgraves.org/2009/05/ ··· ta-rate/

There are a handful of codecs that do require more bandwidth on the wire, but these are typically intended for use with applications beyond voice. That is, they generate full-bandwidth or stereo encoded streams.

Thanks for the information, much appreciated.

The area remains murky in my mind, however:

1) Commentaries such as this:

Narrowband:

* G.711, G.726: Offers narrow-band voice with low processing requirements, and with typical bit-rates of 32-64 kbps.
* Other narrow-band codecs with compression, such as G.729AB and G.723.1, offer bit-rates as low as 5.3 kbps, but require higher DSP processing horsepower.

Wideband:

* G.722: Offers wideband voice with low DSP processing requirement similar to standard G.726 ADPCM. However, it requires a bit-rate of 48-64 kbps per channel.

Brent Lorenz is the IP Phone Product Manager at Texas Instruments.
»www.eetimes.com/design/s ··· P-codecs

2) I've been told similar things by two different engineers.

3) Even saying that G.722 and G.711 require similar bandwidth usage, the fact remains that G.729 uses far less, and thus many VoSP prefer G.729 for exactly this reason.

4) G.722 is royalty-free. That being the case, and if it is not a bandwidth hog, and if it sounds great, then why do so many Voip providers, and so many manufacturers, not support it? In other words, why has adoption been so slow?

I am truly not trying to be difficult, but this does not seem totally clear-cut.

Thanks again, and many thanks for all your reviews on the Gigaset and other IP phones.

MartinM responded with sarcasm first.

neener neener

Hi PX.

You have to love professional standards. There are so many to choose from.

It's my understanding that the Classic flavor of G.722 (as opposed to some of the newer flavors such as G.722.2) runs at 64 kbp/s. That's exactly the same data rate as G.711 (64 kbp/s) but more than all the different flavors of G.726 (16, 24, 32, and 40 kbit/s).

If you're going to make comparisons between codecs please try to compare like-for-like:

G.711 is most closely comparable to G.722, both typically use 64 kbps.

G.729 is most closely comparable to AMR-WB (12.65 -23.85 kbps) Both are patent protected, requiring licenses.

There are similar comparators for g.723, G.726. The simple fact remains that there are wideband codecs in the very same range of bit rates as all the narrowband legacy codecs.

There are in fact a large number of wideband codecs, many of which are not yet broadly in use.

Wideband audio need not consume any more bandwidth on the wire than legacy pstn audio.

WB: AMR-WB (aka G.722.2) 6.6 - 23.85 kbps

WB: AMR-WB+ 5.2 - 48 kbps

NB: G.711 64 kbps

WB: G.711.1 64 - 96 kbps

WB: G.719 32 - 128 kbps

WB: G.722 48 - 64 kbps

WB: G.722.1 24 - 32 kbps

WB: G.722.1 Annex C 32 kbps

NB: G.726 16 - 40 kbps

NB: G.729 8 kbps

WB: G.729.1 8 - 32 kbps

You pose a good many questions, although the answers may not be clear cut. There's a lot of politics in this. That said, I'll try to gather some of it in a blog post later in the week.

While that is true, it kind of doesn't matter.

The new economics of the voice world are based on the current technology, not based on what they were stuck with in the past. Companies are setting their rates based on what it costs them to do business today, and they are competing with other companies that are using today's technology.

You presume a homogenous network, which is simply not the case. There remains large amounts of TDM equipment installed and in-use. The world is not yet completely IP based.

»files.shareholder.com/do ··· 0831.pdf

According to 8x8's recent report to shareholders in the US IP telephony represents only $1.6B of revenue whereas circuit switch business telephony remains $40B.

From “2008-2013 Sizing and Share for Wireline Voice and Data: The Broadband & Business
Boom,” 2008 Atlantic ACM

So it would seem that businesses often compete in today's market using yesterday's technology.

Further, the wholesale cost of bandwidth has fallen consistently over time. Some here have quoted it as $1/mbit/mo.

Since it uses similar data rates to existing encoding schemes, and with the cost of bandwidth decreasing, there can be no argument that deployment of wideband voice is being restrained by issues of bandwidth requirements.

The argument was being made in general terms, as the broad complaint, not the start of this thread, was the HDVoice requires more bandwidth. This is largely not the case.

HDVoice is a relatively new phenomenon in the business. Providers like voip.ms likely didn't architect for HDVoice, it just happens to be possible in some cases because they are leveraging a SIP/IAX2 over IP infrastructure.

When some people find that it's not always possibly for whatever reason they are surprised. However, voip.ms major selling point is cost...their FAQ page doesn't even claim support for any HDVoice codecs.

»voip.ms/faq.php#whatcodecs

Well, it requires more bandwidth than non-HD voice, which is what most of the competition in the voip space is doing.

Anyway, I do understand what you're saying, I just don't entirely agree.

No. It doesn't. Or at least, it needn't. That was the point which you fail to see.

If you are determined to find the absolute cheapest route to whereever then it's outside of you scope anyway.

I accept that many people care more about having it cheap than a high quality call.

In that case I don't follow your argument.

Using today's best technology, HD voice requires more bandwidth than non-HD voice.

Market participants who are competing in the same space are going to use the best technology they can.

The ones who use non-HD voice are going to have lower operating costs (due to the expense of bandwidth and CPU required to process HD voice). They will therefore be able to charge consumers less.

If you are arguing that HD voice with current technology uses the same bandwidth as non-HD voice on old technology, I don't really follow the relevance. The companies in a position to offer HD voice are not competing most directly with the antique players; they are competing with other high-tech upstarts.

It's like you're saying that Hummer H3s are more fuel-efficient than Studebakers and therefore they are an ideal offering in today's high-oil-price environment.

The HD voice codec - G.722 uses a kind of ADPCM compression called sub-band ADPCM, and has selectable 48, 56 and 64 kbit/s payload data rates. Typically 64 is used, which is the same payload size as G.711, and thus same total bandwidth.

Why compare it to g.711 when there are 8 and 16kbps alternatives? We're back to the Studebaker and the H3.