atcomsystems.ca/forum Forums General General Telephone & Systems Information Acceptance of the decline of voice quality

My head is hurting trying to figure out what AFAICT and IFAIK mean.

I just installed a Cisco VoIP. About 30 phones. I really REALLY liked the speakerphone on the conference table. Crystal clear. Picked up the boss talking at normal volume from 10-12 feet away from the mike. HOWEVER.....

A circuit switched 48 channels is carried on a pair of T1's -- 3.088 Mbps. For this VoIP system I installed a pair of D3's. The second is for reduncency, but each D3 is 45 Mbps. Which is cool since one went down the day after it was turned up. I have 15 times the bandwidth leaving as I would require for circuit based telephony.

As Far As I Can Tell... and As Far As I Know the other has a a typo.

FWIW I've been communicating via computers since 1981. YMMV.

Since it's correct or question Mike day... calling a device in a phone that hears the person speaking is a mic not a mike. Short for microphone. Many times the speaker is also the mic. Speakers and microphones are bi-directional transducers. That is, they convert mechanical motion into an electrical signal (a mic) or electrical signal to mechanical motion (a speaker).

Richard

While the phone guy in me agrees with hbiss on the "if it ain't broke don't fix it". I think Mike nailed it. Profit. Any industry is going to go where the money is or they will go the way of the proverbial buggy whip manufacturer.
Allow me to throw these ideas out there and see what you guys think. It appears these are the factors at work.
1. Carriers are convinced they can transmit voice traffic more efficiently on IP. Which can/should lead to lower prices for the consumer and higher profits.
2. Phone system manufacturers can make a system that handles calls more efficiently (more calls per hour) and greater interoperability with other systems.
3. Phone system manufacturers create a huge new demand for IP products.

Richard

Richard Richard Richard

You should have known that was coming :toast:

Voice quality, in the Cisco VoIP system I just installed, is excelent. I also have bandwidth to burn.

My thoughts, correct as needed:

VoIP only requires bandwidth when people are talking.

Circuit switched voice is 64Kbps fixed.

Packet switched voice can be designed to use twice that, for much higher quality; while using less average bandwidth, for lower costs.

Other faults can be corrected with suitable voice processing. I remember playing years ago with VoIP and noticed a slight clipping at the start of any sending. A buffer on the transmitter's part that starts sending slightly before the trigger volume would stop that.

At some point in the near future the switch to switch channels will be migrated to VoIP -- for the cost savings.

Us Technicians and Engineers *want* the higher quality. It's the bean counters we have to fight.

By switch to switch do you mean inter office trunking? If so I do not see that changing much, since VOIP still uses ISDN ckt's (PRI) which is what is used to connect offices now. From a local switch standpoint Voip is nothing more then a PRI

Caveat: I am a Tech, not an engineer. The below is just what I have picked up over the decades:

Switch to switch, interoffice, yes. VoIP uses ISDN, I don't think so.

ISDN is a digital loop service, the BRI is 140Kbps, the PRI is an 1960's vintage T1 repackaged. It is a part of the Circuit Switched Network. Interoffice trunking is the major part of the Circuit Switched Network.

Other than dial-up access, the Internet, AKA the Packet Switched Network, is, as far as I know, completely separate from the Switched Network. VoIP is carried over the Internet to your telephone service provider. The internet is carried in the ATM cloud -- which describes my knowledge of it -- cloudy. You don't use VoIP over a dialup into the Internet, thus VoIP never touches the switched network until it gets to the VoIP provider who connects it to the switched network. If it ever does.