Audio over IP 1

[hhallett]

Does anyone have experience with AoIP? I'm looking for options to ISDN for radio stations to use for remote broadcasts from this hotel. AoIP is suggested as a solution but I don't know enough about it to know what questions to ask vendors and equipment sellers.

Respectfully.
Harry Hallett
Telecommunication Systems Technician
The Queen Mary Hotel

 

[intrigrant]

AoIP = VoIP

 

[sbcsu]

ISDN (Integrated Services Digital Network) or It Still Does Nothing...
Was and probably still is the benchmark for audio over distance for radio stations.
Depending on your territory you had 56K or 64K per channel, you could bond at least 2 channels and it gave perfectly clear audio. The associated data was sent over a separate 9K data channel.
Under set adapters could also be used on proprietary phones to achieve this if no direct ISDN was available.
However we are moving away from Circuit Switched to Packet Switched connections.
You can keep Circuit Switched ISDN but it will become more redundant as time goes on and more expensive.

Packet Switched connections are now Voice over Internet Protocol (VoIP)
This mostly means SIP - Session Initiated Protocol
For you it will mean reading up on Codecs - the digital encoding and decoding of the audio.
Different codecs will give different audio quality.
G703 & GSM would be examples of codecs.
Jitter - just like noise on older circuits.
Bandwidth - not always guaranteed unlike ISDN
Packet Prioritization - Adding a 'header packet' to outgoing voice packets to enable it to transverse the network quickly.
(End users will wait for an email but can not wait for time urgent voice packets)
Signalling Prioritization - Contacting a distant location needs to be prioritized before emails etc. but after voice packets.

Most telephone audio is now packet switched, the only circuit switched really is the telephone line from your premises still running on the now very old 50V DC.
If we were to install a telephone network today it would be a bus between buildings.
After the telephone exchanges our voice packets are transferred into mostly glass fiber to be sent to the destination...

 

[Westi]

also to remember is that phone lines (including VoIP on g711 codec) are only using 3100 Hz (300 - 3400Hz) of the voice spectrum so your sound will not have the entire voice spectrum when you use a regular phone line.
The codec g722 uses a higher spectrum (50 - 7000Hz) but needs to be used on both ends, and can't be converted between source and destinatio to save on bandwidth, to transmit the additional spectrum.

I am sure that there are applications out there that will use a decent amount of bandwidth and transmit better audio quality. If you are using it for AM radio you don't need to care but if you need it for FM radio then you would hear the depreciated spectrum and it would sound like a phone call.

I have had phone calls with g722 and it sound clearer and more defined when you talk to someone, it also makes a difference what voice is on the other end as a deep male voice doesn't sound much different than a higher male or female voice.

Joe W.

FHandw, ACSS (SME)


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