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Answering A Few Questions About HDVoice

Some time ago over in the VOIP Forum at Broadband Reports there was an interesting thread about wideband telephony and it’s relationship to common ITSPs. The person who started the thread was surprised to find that they could not pass wideband audio between two accounts on the same SIP-based ITSP.


Holy cow, I just tried to pass G722 and Speex wideband through and it choked !! I can’t believe it, I thought they were supposed to be one of the more competent providers.

This person’s presumption was that if the call is between accounts at the same service provider it doesn’t traverse a media gateway, and so should support any desired codec. While the logic was basically sound, clearly this wasn’t the case as the person soon found out.

The thread goes on to get quite involved, including mentions if SIP Proxies, DNS, ENUM, calling via SIP URI and various codecs. It also highlights certain happy coincidences, for example;


Using Callcentric, if I call my other Callcentric line, it uses the g722. From my account, the g722 is moved to g711.

This participant has noticed that a particular service provider doesn’t proxy the media between on-net calls. Thus people with accounts at that company can call each other using G.722 and enjoy a wideband call. This just happens to be possible even though the company makes no specific claim to be wideband capable. It just happens that their SIP architecture supports the use of an arbitrary codec between on-net end-points.

The thread goes further, and some people mention the longstanding myth that wideband voice consumes more bandwidth on the network than traditional, narrowband telephony. This is a matter that I’ve already addressed several times, most notably here.

The thread then degenerates a bit, influenced by participant who constantly asserts that HDVoice must require more bandwidth, in spite of clear evidence to the contrary.

However, one Broadband Reports regular asks a couple of good and concise questions, which I promised to try and address.

PX Eliezer

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.

It’s true that some carriers prefer G.729 over G.711. These are typically long haul carriers trying to achieve the cheapest possible routes to otherwise expensive destinations. That is to say, their primary concern is not for quality but price.

Even so, as I have said before in one form or another, there are comparable wideband-capable low-bitrate codecs to just about everything in the narrowband PSTN realm. G.722 can only rationally be compared to G.711…which both share the same core data rate.

If you’re considering G.729a or G.723 then you should be comparing them to AMR-WB, aka G.722.2. G.729a at 8 kbps is actually a little lower data rate than AMR-WB at 12 kbps, but the audio quality difference is starting. The major trouble is that both are patent protected and so incur significant licensing fees.

My turn to pose a question: If you’re building out a new network do you choose to simply add network capacity? Or opt to pay the license for a patented codec to squeeze more calls into less network capacity? What’s the more future-proof strategy?

G.729a was developed at a time when networks were nowhere near as capable as they are today. Thus trading off license cost vs network build-out cost made some sense.

That was 20+ years ago. That same situation may not make the same sense today. Wholesale bandwidth is dramatically cheaper today, some have told me $1/mbps!. This makes it less practical to pay a high per-channel licensing cost for the compression scheme, at least on wired networks.

Context is important in find the logic of the situation. HDVoice is a trend that is happening as we move into the future. It’s forward looking. Further use of G.729a on wired networks is more about wringing value from the past. It’s a means to cram more calls into an existing network and hopefully avoid any new network build-out. It’s at best a stalling strategy.

Cellular providers live in quite a different world. To them conserving over-the-air bandwidth is worth the price of the proprietary codec license fees. Thus 3GPP included AMR-WB (typically 12.65 kbps) as their standard wideband audio codec for mobile networks. Similarly, those who are proponents of CDMA networks (Qualcomm, Sprint & Verizon) adopted EVRC-WB as their standard HD codec. It cruises along at 8.55 kbps, but it’s also patent encumbered.

If you want more info about codecs check out this Polycom white paper; VoIP to 20 kHz: Codec Choices for High Definition Voice Telephony.

That’s his first question. I’ll return to address his second question later in the week.

This Post Has 5 Comments
  1. Michael,

    The Asterisk 1.8 released few days ago. It seems to be supporting HDVoice, isn’t it?

    1. Asterisk has supported HDVoice in the form of G.722 for well over a year. Support for G.722.1 and G.722.1C were added in the v1.6 release. I’m sure we’ll hear more about all the wondrous new features of v1.8 as Astricon progresses this week.

  2. *sigh* I’m working to get out the HD Voice whitepaper by end of the year.

    Mobile HD voice has got a head of steam; Egypt (Cairo), India (Multiple zones), and Russia (Moscow) have all gotten onto the HD voice bandwagon in the last two weeks…

  3. I was someone who thought HD/wideband was just a silly marketing thing until last night and now I can’t wait until the most basic cell phone can make HD calls.

    Recently I signed up for onsip (will probably just use it as SIP call provider) and i registered my yealink tp22 to their network. I know my yealik is not the most ideal phone and its one that you certainly let in your house but I’m working on deploying it at a dentist office. its not too bad of a phone and for $75, it works really well.

    Anyway, the tp22 was registered to onsip and I have a free softphone one my samsung galaxy 2 and I called in HD over wifi. This was just a silly test since I was just talking to myself but it really did sound SO GOOD!

    Later on, I made a call using the same phone but a different VOIP provider over the PSTN and got the sound quality I’m used to. While on the call, I called on onsip to my softphone and had my wife answer the call on my cell phone. I then conferenced all three of us together using only one PSTN connection. My wife and I sounded very clear to each other, and as expected, the PSTN call was garbage.

    onsip called just a sip2sip address.

    feel free to call me, in HD, friends.

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