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It seems that agere ICs is better.
But I do not know is it necessary to write VOIP protocol stack for these ICs(they are only a hardware platform ) or all the software/hardware protocol stacks is included in the
IC,and I just configure and use it.which is correct?
It seems that agere ICs is better.
But I do not know is it necessary to write VOIP protocol stack for these ICs(they are only a hardware platform ) or all the software/hardware protocol stacks is included in the
IC,and I just configure and use it.which is correct?
Generally all chip manufacturer presents hardware reference design and
reference softvare or software SDK, but sometimes it is very costly 5-40K USD .... for mass production it is ok, but for small serries production ....
Yea, went down this road with TI's VOIP solution. The target price per device is very inexpensive (~5 per unit with IP indemdification) but you are tied into the telogy solution. That is, you use the VOIP solution as they provide it and you cannot add or change the internal stack and codec funtionality. The DSP you get runs a pre-compiled solution from Telogy only. So, essentially you are getting an entire VOIP phone solution but if you wanted to do something unique, you are out of luck....
Telogy was very adament about "you have to do it our way" at which point some of us raised our middle fingers.
And yes, there is an upfront "to get started" development fee for around 30K USD.
If I want to send/receive voide over a LAN (ethernet 100Mbps)is there any alternative for VOIP? Can I use a simpler way in embedded?
Only voice comm is used for local network and all devices for voice comm will be made by my design.
I think for a solution to get data from a codec chip and send it over a tcp/ip network by a tcp/ip packeting ,The protocol used does not need to be standard like as VOIP.
There are several open source VOIP stacks (such as Voivida) available with at least a PCM codec. It just means that it might take a little work to port it to your specific platform. Where as, the Telogy solution is a "ready to implement" solution.
I would like to build a VoIP intercom.
Two identical stations at different locations within a LAN or WAN.
Just connect power, a PC speaker & mic and Ethernet. TX activated by a PTT switch. I/P address and other parameters configurable by a terminal via RS-232 serial port. Doesn't have to be full duplex so no echo cancellation is required.
I'm wondering if it would be possible to do using an Atmel ATmega processor with a suitable hardware CODEC and a Realtek 8019as NIC chip ?
Would an ATmega chip have enough horsepower (MIPs) to handle reading the CODEC and dumping UDP packets and keep up ?
Anyone done anything similar to this ? Ideas, suggestions etc. appreciated.
(Looking for homebrewed project ideas, not commercial VoIP phone solutions.)
If you intend to use it on LAN as P2P connection with no other nodes interfering then you can just use G.711 PCm codecs and send raw packets on layer2...no need for implementing UDP...also no jitter to be expected (o;
You can even use ethernet wiring for supplying power supply to endstation...
For performance figures...
68k Motorola @ 16MHz running Linux stack can keep up easily with receiving 192kBits/sec Multicast mp3 stream and pipe to mp3 chip (o;
Hi
You must understand all the RPT stuff first
The lowest end is to use a PIC with an MT9161 codec, you can
use the pic's xout to drive the clock and do assync frames.
Another real nice chip is the **broken link removed** from OKI.
If you want sw, get full stuff on openh323.
Best regards
Look at www.asterisk.org
It is a full functioning ip pbx ...can do a intercomm aslo
radeohedca said:
I would like to build a VoIP intercom.
Two identical stations at different locations within a LAN or WAN.
Just connect power, a PC speaker & mic and Ethernet. TX activated by a PTT switch. I/P address and other parameters configurable by a terminal via RS-232 serial port. Doesn't have to be full duplex so no echo cancellation is required.
I'm wondering if it would be possible to do using an Atmel ATmega processor with a suitable hardware CODEC and a Realtek 8019as NIC chip ?
Would an ATmega chip have enough horsepower (MIPs) to handle reading the CODEC and dumping UDP packets and keep up ?
Anyone done anything similar to this ? Ideas, suggestions etc. appreciated.
(Looking for homebrewed project ideas, not commercial VoIP phone solutions.)
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