VK2RK
Active member
Amateur Radio Voice linking protocols
Preamble
Back in 1995 a voice transmission system was developed that allowed the transmission of sound via the internet, Voice Over Internet Protocol was born (VOIP) initially used largely in telephony reducing the cost of calls to the end user.
With development of system a switching system was created called Asterisk, this served in the role of an exchange and interface to the phone switched network.
The VOIP protocol provided several encoding systems that provided degrees in audio quality at the cost of bandwidth.
For more detailed information on the structure of VOIP
Audio codecs
G.711
Introduced in 1988, G.711—the international standard for encoding telephone audio on a 64-kbps channel—is the simplest standard among the options presented here. The only compression used in G.711 is companding (using either the µ-law or A-law standards), which compresses each data sample to an 8-bit word, yielding an output bit rate of 64 kbps. The H.323 standard specifies that G.711 must be present as a baseline for voice communication.
G.723.1
G.723.1 is an algebraic code-excited linear-prediction (ACELP)-based dual-bit-rate codec, released in 1996 to target VoIP applications. The encoding time frame for G.723.1 is 30 ms. Each frame can be encoded in 20 bytes or 24 bytes, thus translating to 5.3-kbps or 6.3-kbps streams, respectively. The bit rates can be effectively reduced through voice-activity detection and comfort-noise generation. The codec offers good immunity against network imperfections—like lost frames and bit errors. G.723.1 is suitable for video-conferencing applications, as described by the H.324 family of G.729
Another speech codec, released in 1996, is the low-latency G.729 audio data-compression algorithm, which partitions speech into 10 ms frames. It uses an algorithm called conjugate-structure ACELP (CS-ACELP). G.729 compresses 16-bit signals sampled at 8 kHz via 10 ms frames into a standard bit rate of 8 kbps, but it also supports 6.4 kbps and 11.8 kbps rates. In addition, it supports voice-activity detection and comfort-noise generation.
VOIP Technology as used in Amateur Radio
Sometime after the introduction of VOIP in 2002 Jonathan Taylor develop Echolink this is basically a VOIP system with a modified control layer that allowed a point to point connection or a group connection as in a star topology.
With this the use of a radio was not necessary, one could connect to another amateur just using a computer
The system is still used today, its advantages are:
a) Ease of use
b) Easily interfaced into a radio
Disadvantages are:
a) Poor security
b) Audio quality only as good as the user implementation
IRLP (Internet Radio Linking Protocol)
https://en.wikipedia.org/wiki/Internet_Radio_Linking_Project
Very early in the development of the technology for Amateur Radio, Dave Cameron (VE7LTD) created the IRLP system This offered a Peer to Peer secure connection, later offering the ability to have a star topology allowing several nodes to be part of a local area network.
Technically IRLP can be used to communicate just using computer to computer as Echolink does, however it was from the very start designed to be part of an RF radio system.
It provided a very secure environment as it uses a unique user key that stops any traffic unless this key is known to the system.
The Audio quality is very good, mainly due to the restriction placed on home built interfaces that only a device built by VE7LTD is allowed on the network, with this audio quality is maintained in the network.
Allstar
The late comer in the technology (2018) is the protocol Allstar, largely born to counter some of the security issues with Echolink providing either Radio or Computer connection and linking.
The only disadvantaged is that any station can connect to any other group of stations bringing those connected to it in to the group, so the control of connections aside banning users is not acceptable for radio repeater to radio repeater linking as the IRLP system offers.
Conclusion
In all of the above modes VOIP CODECS are used with the aim on audio quality and allowed bandwidth.
Can these systems be called digital ? No they are analog in analog out at either a computer or radio, but the information transport is digital using packets in the VOIP protocol.
When any of the above systems are used, one must consider the security aspect and the kind of traffic to serve. Presently IRLP offers the best in security and audio quality.
Preamble
Back in 1995 a voice transmission system was developed that allowed the transmission of sound via the internet, Voice Over Internet Protocol was born (VOIP) initially used largely in telephony reducing the cost of calls to the end user.
With development of system a switching system was created called Asterisk, this served in the role of an exchange and interface to the phone switched network.
The VOIP protocol provided several encoding systems that provided degrees in audio quality at the cost of bandwidth.
For more detailed information on the structure of VOIP
Audio codecs
G.711
Introduced in 1988, G.711—the international standard for encoding telephone audio on a 64-kbps channel—is the simplest standard among the options presented here. The only compression used in G.711 is companding (using either the µ-law or A-law standards), which compresses each data sample to an 8-bit word, yielding an output bit rate of 64 kbps. The H.323 standard specifies that G.711 must be present as a baseline for voice communication.
G.723.1
G.723.1 is an algebraic code-excited linear-prediction (ACELP)-based dual-bit-rate codec, released in 1996 to target VoIP applications. The encoding time frame for G.723.1 is 30 ms. Each frame can be encoded in 20 bytes or 24 bytes, thus translating to 5.3-kbps or 6.3-kbps streams, respectively. The bit rates can be effectively reduced through voice-activity detection and comfort-noise generation. The codec offers good immunity against network imperfections—like lost frames and bit errors. G.723.1 is suitable for video-conferencing applications, as described by the H.324 family of G.729
Another speech codec, released in 1996, is the low-latency G.729 audio data-compression algorithm, which partitions speech into 10 ms frames. It uses an algorithm called conjugate-structure ACELP (CS-ACELP). G.729 compresses 16-bit signals sampled at 8 kHz via 10 ms frames into a standard bit rate of 8 kbps, but it also supports 6.4 kbps and 11.8 kbps rates. In addition, it supports voice-activity detection and comfort-noise generation.
VOIP Technology as used in Amateur Radio
Sometime after the introduction of VOIP in 2002 Jonathan Taylor develop Echolink this is basically a VOIP system with a modified control layer that allowed a point to point connection or a group connection as in a star topology.
With this the use of a radio was not necessary, one could connect to another amateur just using a computer
The system is still used today, its advantages are:
a) Ease of use
b) Easily interfaced into a radio
Disadvantages are:
a) Poor security
b) Audio quality only as good as the user implementation
IRLP (Internet Radio Linking Protocol)
https://en.wikipedia.org/wiki/Internet_Radio_Linking_Project
Very early in the development of the technology for Amateur Radio, Dave Cameron (VE7LTD) created the IRLP system This offered a Peer to Peer secure connection, later offering the ability to have a star topology allowing several nodes to be part of a local area network.
Technically IRLP can be used to communicate just using computer to computer as Echolink does, however it was from the very start designed to be part of an RF radio system.
It provided a very secure environment as it uses a unique user key that stops any traffic unless this key is known to the system.
The Audio quality is very good, mainly due to the restriction placed on home built interfaces that only a device built by VE7LTD is allowed on the network, with this audio quality is maintained in the network.
Allstar
The late comer in the technology (2018) is the protocol Allstar, largely born to counter some of the security issues with Echolink providing either Radio or Computer connection and linking.
The only disadvantaged is that any station can connect to any other group of stations bringing those connected to it in to the group, so the control of connections aside banning users is not acceptable for radio repeater to radio repeater linking as the IRLP system offers.
Conclusion
In all of the above modes VOIP CODECS are used with the aim on audio quality and allowed bandwidth.
Can these systems be called digital ? No they are analog in analog out at either a computer or radio, but the information transport is digital using packets in the VOIP protocol.
When any of the above systems are used, one must consider the security aspect and the kind of traffic to serve. Presently IRLP offers the best in security and audio quality.
Last edited: