Getting ready for VoIP

 In Audio, Blog

tincan_400-e1433784355229I’ve heard many times from integration partners that they prefer to use analog telephone lines vs. VoIP lines for their conference room installs because it just works.

While I totally understand the ‘familiarity factor’ of using analog twisted pair for telephony and knowing it will work like the hundreds of other times you’ve integrated with analog telephone lines, I think there’s an aspect of ‘kicking the can down the road’ that should be addressed for the day when there will be no more analog telephone line installs. To make it easier when that day comes I’m going to try to increase the comfort factor for VoIP sooner rather than later.

VoIP (Voice over IP), like every other new topic, is not as intimidating once you learn more about it and start building a base of knowledge.

I’ll start with SIP, Session Initiation Protocol, as it is one of the most common protocols used for VoIP. SIP is basically a way of finding other SIP users, sharing what capabilities you have for communication, and starting and ending communication sessions. It’s a text-based protocol (so you can see the messages on the wire) that’s well-defined so that manufacturers’ equipment can talk to each other.

What are some of the big items to keep in mind about SIP vs. Analog Telephony? We’ll focus on a couple here and have more in future articles.

Line Registration

SIP operates with the phone or telephony interface (also called a user agent) registering to a call server. A call server is what we used to call a PBX. This registration is for a particular line – think of a line as an ‘extension’ off the PBX that can be used to initiate outgoing calls and receive incoming calls. To register this line with the call server there are some authentication credentials (i.e., user name and password) that must be entered both on the call server and on the user agent. This is where the integration partner needs to communicate with the customer’s local IT team to get the right credentials and put them into the right place on the user agent’s configuration software. To see where to enter the credentials on the telephony interface, follow your DSP manufacturer’s training and instruction. Yes, that means reading the manual.

Dialing

VoIP interfaces can be dialed like analog telephony with off hook dialing or like a cell phone with on hook dialing. Off hook dialing means you take the phone off hook and start pressing digits until you’ve entered enough digits for the call server to dial a valid number. On hook dialing means you dial some digits and take the phone off hook which, in turn, tells the call server to dial those digits.

The only trick with off hook dialing is how does the call server know you’ve dialed enough digits for a valid number? That’s where the digitmap comes in to play. More on that in a future post.

Audio codecs

Once you have a registered line and know how to dial calls, it’s helpful to understand how audio from one side of the call gets to the other end. With analog telephony the call was either analog (in the old days) or compressed to 64kbps using G.711 u-law audio compression and sent digitally. The quality of the call using 64kbps G.711 u-law compression was called ‘toll quality’, which meant it sounded as good as an older analog call. With VoIP there are many more options for speech compression from wideband G.722 (higher than toll quality) to very low bit rate speech coders such as G.723.1 at 5.3 kbps which is lower than toll quality. If you have audio fidelity issues in your calls, check with speech coder is being used and look to use more bandwidth with a higher quality codec.  Yes, that also means you may need to read the manual.

Learning more

I’ll post more information about SIP in future posts. In the mean time if you want to learn more about SIP in a comfortable group environment, Aveo Systems is delivering a technical presentation at Infocomm 2015 entitled ‘De-Mystifying Voice over IP in Collaboration Environments”. This is session IS46 on Thursday June 18 at 12:30pm. Hope to see you there!

For more detailed technical information, see the IETF’s complete SIP specification, RFC3261.

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