VoIP Codecs, Which One Is Right For My Environment? - LinkedIn

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In previous editions David takes on various technology topics and breaks them down into simple terms everyone can understand. Most recently he talks about the most widely used VoIP Codecs, and a few you've never heard of...

"We’ve talked a lot about SIP and H.323 and the differences between the two. We’ve also talked about how these two protocols are just signaling for something else. Well, when it comes to telephony, they are both signaling for some form of voice stream generally referred to as VoIP (ie Voice over IP). I generally think of this as the bearer path portion of the phone call. At a very simple level, we are converting sound into packets of data to be sent onto the data network. Then once they reach the destination, they need to be converted back into voice for the other person to hear. This “coding” and “decoding” has a name, conveniently known as a codec. The interesting part about Voice over IP is that there are a lot of ideas as to how this should be done. VoIP equipment, such as PBX’s, phones, gateways, media servers, etc., have a lot of choices when it comes to Codecs. This week I’d like to talk about a few of those voice codecs that we use a lot.

For starters, know that each codec choice has advantages and disadvantages. There is usually a tradeoff between the quality of the conversion and the amount of bandwidth it takes up on the network. It is important to know that the codecs themselves are very standard and have very defined bandwidth amounts. For example, one very common codec is G.711. Every manufacturer seems to support this codec because it is the one that is meant to replicate the exact algorithm used in TDM environments, specifically those used in the PSTN (Public Switched Telephone Network). It uses 64kbps. But keep in mind that this is just the payload of the packet. As that payload works down the OSI model to get the onto the data network, I have to encapsulate it in more and more layers of overhead. In the end, the type of data network will determine the amount of overhead required. So typically, a G.711 call over a typical data T1 (ie PPP) network ends up taking up about 82kbps. Fortunately, the data guys can implement RTP Header Compression and drop that back down to about 67kbps. My favorite bandwidth calculator to use is https://www.bandcalc.com/, as it does a great job of showing you each layer of encapsulation and what overhead gets added at each.

G.711

As I mentioned, this is an extremely common codec. When you want it to sound exactly like it did in a TDM environment, this is the codec to use. This was made a standard in 1972. It uses the same algorithm that is used in regular TDM telephony calls. This means the same Pulse Coded Modulation, using 8 bit non-uniform quantization with 8,000 samples per second. Blah, Blah, blah. It means “Toll Quality Voice”. As stated above, it uses about 67kbps over the PPP T1 (with compressed RTP Headers). It has traditionally been the gold standard upon which all other codecs are compared.

G.729

This was the next one to become popular. There are a lot of flavors, called Annexes, of this one. You see these as G.729A or G.729B, etc. But the general idea is to use less bandwidth, usually at the cost of voice quality. For example G.729A uses ACELP (Algebraic Code Excited Linear Prediction). You don’t even have to know how it works, and you’ll sound really smart by being able to rattle that one off. Its payload uses about 8 kbps, but with encapsulation overhead, you’re at about 11kbps of bandwidth (assuming a compressed RTP Header on a PPP T1). This allows you to get more than 6 times the number of phone calls over the same WAN connection as you would with an uncompressed codec like G.711. Now, G.729 Annex B (or simply G.729B) is a little different. It prefers to use Silence Suppression as its way to reduce bandwidth. I’m NOT a fan of Silence Suppression at all!! No vendor implements it well. Silence Suppression is a technique that takes advantage of the fact that humans don’t talk in full duplex. One person in a conversation generally talks, the other listens. So, instead of transmitting bandwidth during periods of listening, Silence Suppression first detects the silence and then tells the receiver of the silence to simply play local white noise. Without this white noise, we are trained to think the system isn't working. The White noise makes us feel more comfortable with what's happening. The problem is those crazy soft talkers (remember that great Seinfeld episode when Jerry has to wear his soft talking, designer girlfriend's puffy pirate shirt on the Today Show because no one could understand what she was saying?). Well, even the VoIP system has a hard time with that. The system has a hard time telling the difference between a very quiet voice and simple background noise. The end result can be horrible clipping of voice. I hate silence suppression! So, I hate G.729B. Don't get stuck having to wear a puffy pirate shirt!!! To be honest, I’m not a fan of any of the G.729 flavors. I think it is overly compressed and sounds bad.

G.726

G.726 is my favorite of the compressed codecs. Relatively new (ie 1990), it uses ADPCM (Adaptive Differential Pulse Code Modulation). This codec is based very much on the same stuff we talked about with G.711. But that “Adaptive Differential” part lets me reduce the bandwidth significantly. It’s not as low as G.729, but to me, sounds just as good as G.711, with only 35kbps over that typical compressed header data T1. Again, I can’t tell the difference.

G.722

The next codec I’m going to talk about takes a very different approach. Besides the “original” G.711, most of the other codecs tend to try to compress the call, lowering voice quality and bandwidth. G.722 takes a different approach. It’s asks the question, “Why are we trying to make it sound worse? Who said Toll Quality Voice should be the gold standard?” G.722’s goal is to make it sound better! Instead of taking 8,000 samples per second like the other guys, it takes 16,000 samples per second with a wider tonal range, resulting in significantly better voice quality. Normally, this would take twice as much bandwidth, but because it uses the ADPCM that we talked about with G.726, it only needs about half of the original. This puts us right back at the 64kbps original payload of G.711. Yes, G.722 is awesome! There are some variations of G.722, but they really aren’t the Annexes that we talked about it G.729. These are typically patented and used in more proprietary ways, and need to be licensed from that manufacturer. Some manufacturers will pay for this licensing themselves and simply bake it into the cost of their own licenses. I’m finding that most end users are expecting this type of voice quality. Back in the old days cell phones lowered the bar for us. We could always say “Sure, our voice quality is bad, but at least we sound better than cell phones!” Well, those days are over. All the Apple and Android leverage the carriers’ use of high bandwidth data (such as LTE), and mostly use VoLTE (Voice over LTE). So, this where we don’t really even use cellular minutes anymore. Even our voice calls are all using VoIP based streaming now. And since they all have so much more bandwidth to work with, they’re all using HD audio. You can REALLY tell the difference. Now, PSTN calls are what sound horrible, when compared to cell phones. So, I’m always recommending that customers step up their game, and use these wideband HD audio codecs, such as G.722, as their first choice in their codec lists.

A lot of vendors are starting to embed support for Opus into their endpoints, media servers, and DSP resources. Opus was designed from the ground up to be used for streaming audio over the Internet. It has amazing ability to tolerate much higher levels of packet loss and jitter that we can often run into with uncontrolled networks like the Internet. If you’re interested in learning more about Opus or any of the codecs that we talked about here today, contact ConvergeOne and one of our Solution Architects, Design Engineers, and Implementation Engineers can help you with some current best practices around codec selection and bandwidth utilization."

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Share 3 1 Comment Donald Samuel Kallberg 1y

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0 comments!! Unbelievable!! This is really useful information Thanks Currently using PCMU. Is there any advantage of PCMU over G722? Best regards Samuel in Malta

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