voice codec for VIC cards

[abhishekbhat]

What is the deafault voice codec used when you configure a FXS<FXO or E&M card on a router. Is it constant and if it changes on what does it depend? I know you can change it but I am interested in teh default settings.

[venkekri]

There is no codec concept on the analog side. Its always 64kb/s which g711. Codec comes to play when one is on the wan side or in other words Voip.

Hope that answers your question

[abhishekbhat]

Thanks for that insight but it still leaves me in doubt as to the codec on the WAN side. What dictates the codec on the WAN side because if you do a 'show voice port' or 'show voice dsp' the router output does show a codec. What is that?

[pacameron]

An analogue voice port must be able to convert the signal from the FXS, FXO or E&M port into a digital bit dtream for the DSP to process. So each analogue voice card has at least 2 dedicated codec chips that do this conversion of the analogue signal to a 64Kbps bit stream. By default, they output G711ulaw.

The DSP takes this bit stream and then processes it according to the configured codec under the VOIP dial peer. Depending on the configured codec, all the DSP does is break the bit stream into 20 msec samples, compresses the audio if it is a low bit rate codec then wraps RTP, UDP and IP headers around the sample. Likewise, in the return path , the DSP pulls off all the IP/UDP/RTP headers , decompresses the audio and feeds out a 64Kbps bit stream back to the codec on the voice port, which in turn converts the digital bit stream back to the analogue voice port.

So ... there is a codec to the analogue to digital and digital to analogue conversion on the voice card, and the DSP also does a codec function to convert audio from a 64KBps bit stream to a compressed/packetised format.

Hope this clarifies things.

[abhishekbhat]

In case there is no codec configured under the dial peer mode, does it mean that the output on the WAN utlizes 64K. And does the DSP use any voice codec in this case besides G711?

[pacameron]

No - the default codec is G729 for IOS, so the VOIP bandwidth will be around 24KBps if using PPP across the WAN.

Look at it this way

Audio from voice port ------>DSP----->packets

The audio going from the voice port going into the DSP is a 64KBps bit stream. The voice port uses a seperate Codec to generate this digital bit stream.

The DSP packetises the audio, so it is also doing a codec function. It converts the 64KBps bit stream to packets , which can be the same bit rate , or a lower bit rate.