We handle HD voice in a couple of different ways depending on the type of call taking place.
OnSIP is a hosted SIP PBX service, meaning that it is entirely based on SIP (Session Initiation Protocol). Currently, SIP is the de facto protocol for VoIP communications. To learn more about SIP, visit the links below.
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Calls that go through the Public Switched Telephone Network are by default negotiated on the G.711u standard-quality codec, commonly known as ULAW. Nearly all SIP phones support ULAW so your HD phone will have no trouble communicating with other devices through the PSTN, but the call will not be in HD.
These types of calls include all sessions between two SIP accounts/devices/applications that do not go through the PSTN, whether it's an extension to extension call on your network or a call to or from another SIP phone located on the Internet.
Here is where being a hosted SIP provider has its advantages. OnSIP allows the SIP user agents (deskphones, softphones, mobile SIP agents, etc) involved in the communication session to negotiate the terms of the call with each other. User agents can communicate via any media type or codec that they both support.
For example, if two SIP phones sitting on the internet both support the G.722 wideband codec, then there is literally nothing stopping them from having an HD call. It's the same with video calls, or any other types of shared media formats.
Our HD conferencing service acts a lot like a back-to-back (B2B) user agent. B2B services sit in between calls, setting up individual call legs between each end user agent involved and connecting them together.
If a conferencing service does not support HD voice, then the session will not be in HD even if all the callers involved are using HD devices. Our service enables HD conferencing by taking two necessary steps.
1. It establishes individual HD calls with each of the callers for HD inbound audio.
2. It takes the audio from the calls, converts it all to HD, and sends it back to each phone.