SIP-VoIP Critical Concept – Ignore G.711-G.729 CODEC Asynchronous Transcoding at Your Peril

March 12, 2008
I was a speaker this week at Channel Partners on Microsoft OCS-Office Communications Server and SIP which was followed by a session on SIP Trunking. Surprisingly or maybe not, the panelists explained incorrectly the two major types of CODECS.  So, here’s a technical tutorial on CODECs Digital Signal Processors. That is, if you sell VoIP/SIP you should know when/why to use each CODEC type and what high or low performance you will get when you do. If you buy/implement it, you should know what happens when you get DSP-Digital Signal Processing delays arising from "asynchronous transcoding." 
A CODEC-COder-DECoder (also known as an encoder-decoder and COmpression-DECompression system when used in video systems) is a computer chip (semiconductor) DSP-Digital Signal Processing system. Source codecs are designed specifically for speech, whereas Waveform codecs work well with any type of sound. Depending on the audio or voice application would drive the selection of the Source or Waveform CODEC. While there are many types of CODECs, G.711 (wave form) & G.729 (source form) are the two most-commonly CODECs used in VoIP systems. There are many other types of CODECs used in special applications such as Polycom’s SIREN and RT-A-Real-Time Audio used in Microsoft’s OCS-Office Communications Server.
 
Shown here is a G.711 encoded audio stream is 64/56/48 KBPS-Kilo Bits Per Second. Each 13/14 bit sample of the original signal (voice-audio) is encoded into an eight bit Byte/Octet.   Compression algorithms operate by sampling voice and quantizing the analog sound into digital values. G.711 is based on traditional Nyquist-Shannon sampling theorem that the sampling frequency rate must be at least twice as high as the highest input frequency for the result to closely resemble the original signal. A 4,000 Hz-Hertz voice pattern would be sampled at a rate of 8,000 BPS-Bits Per Second.  
 
Here's the "so what" or "why should I care about this." For example, different CODEC sampling rates may start synchronized but shortly become un-synchronized which can cause encoding problems and voice to jitter. To measure and manage jitter RTP-Real Time Protocol uses the time-stamp function in the protocol to assess jitter based on the delay between arrival (interarrival) times of each packet. Changing the number of bits sampled and quantized can dramatically impact the voice quality.  However, LAN-Local Area Network and WAN-Wide Area Network bandwidth limitations may have an equal or greater impact on VoIP performance. Echo can also occur as a result of Asynchronous Transcoding. Transcoding is the process of conversion between circuit-switched (PSTN-Public Switched Telephone Network) and packet-switched networks such as Frame Relay, IP-Internet Protocol and ATM-Asynchronous Transfer Mode. The point is that Asynchronous Transcoding should be avoided. According to Intel, "The term "asynchronous transcoding" refers to a situation when, for example, one endpoint is talking G.711 to another endpoint talking G.729 or two different encodings)."  The point is that Asynchronous Transcoding should be avoided and if you ignore this issue, your calls may certainly be in peril.
 
Next time – tutorial on bandwidth required for telephone calls in SIP/VOIP.
 
This information is also part of OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. The online version is $299 for SIP 2.0c and for $499 as part of OCS-101 Office Communications Server online version per person or less with discounts. For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com Discounts are also available to members of the SIP Forum. For a complete detailed course outline go to: http://www.techtionary.com/ocs/sip-essentials.htm
 


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