'HD voice' on VoIP calls

Am I right in saying that A&A only offer G.711, which is your standard phone call audio quality?

Is there any reason to want more than this currently? (If the answer to that is no, will the closure of conventional PSTN and move to digital voice change that?)

I have been wondering about the audio quality over the phone network too (is it still called PSTN when everything is VoIP?).

I can understand why AAISP wanted to keep things simple and only support a single standard codec but I am starting to think that the world has moved on now. I have a Gigaset DECT base station - so audio is digital all the way to the handset.

My parents have recently switched to Zen's "Digital Voice" service and they state that they offer "HD-quality calling". But I am not sure what that actually means. For example, does it support the same codecs that VoLTE support, when calling a mobile?

I’m assuming it’s done for simplicity.

As a rule of thumb, everyone supports G.711. Adding additional codecs does add some complexity, both for the operator (AAISP) and the end user.

Issues occur for example if you’ve set your preferred codec to G.722 and the number you are calling has an IVR, but the destination VOIP handset doesn’t support the G.722 codec (but the IVR does). Either something in between must transcode, or the call drops unless mid call codec change is enabled, and you support what’s being asked for.

Also, you’ll have odd DTMF issues too.

Calls between my BT Digital Voice service and EE mobile are definitely a much higher quality than "normal" calls. I presume that a lot of the interop between networks happens outside the VoIP realm which drops the quality of everything down.

The mobile providers have high quality voice working between networks so you'd hope VoIP can follow at some point.

In reply to a post by jpm:

Calls between my BT Digital Voice service and EE mobile are definitely a much higher quality than "normal" calls.

That is really interesting to know, thanks!

I presume that there is a company who provides trunked access the phone network for AAISP - so they would also need to support higher quality codecs.

While the PSTN is active most providers will default to 711 when the calls are going out their network,
Calls within a providers network may be on a better codec like 722 although this is better quality it uses the same bandwidth as 711.
I personally use the OPUS codec online and it sounds as good as 722 but at an eighth of the bandwidth.

Thanks
Dan

In reply to a post by xela:

Am I right in saying that A&A only offer G.711, which is your standard phone call audio quality?

Is there any reason to want more than this currently? (If the answer to that is no, will the closure of conventional PSTN and move to digital voice change that?)

Hello o/

There are a few reasons why we don't support CODECs other than G.711

Mainly:
* We have multiple upstream carriers, and they don't support 'HD' CODECs
* Our SIP platform does not support it, and by design we want to handle the media ourselves but don't want to be transcoding between CODECs - so sticking to G.711 makes sense for this

This may well change in the future though.

Thanks for this - and all the other comments in this thread.

The only time I’ve noticed ‘HD audio’ is on FaceTime Audio calls, and it is a pleasant step up from standard telephone audio quality, so would be nice if A&A supported it at some point.

Give them a call, they're usually really helpful.

I use them already for my voip. actually, the thing that spurred this query was my parents’ irrational attachment to their landline, and me wondering whether ‘HD audio’ might be the thing that encouraged them to switch to VoIP. (Answer: not yet.)