[asterisk-users] Transcoding question

[Jim Dickenson]

On Mar 25, 2010, at 11:26 PM, Jeff Brower wrote:

> Jim-
> 
>>> Jim-
>>> 
>>>> There will be up to 150 phones so there will be 300
>>>> channels when they are all on the phone at one time.
>>>> 
>>>> I will be using a current 1.4 version.
>>> 
>>> That's a lot of channels for Asterisk... IIRC the TC400B transcoding card is rated at up to 96 G729 channels.
>>> 
>>> Can you clarify your recording requirement?  Do you mean you want to record compressed (G729) channels, or
>>> uncompressed (G711 or 16-bit PCM)?  You could try to avoid transcoding by using softphones that can do G729, then
>>> you'd have a pass-thru situation -- but if you need to record uncompressed .wav files, then this wouldn't work.
>>> 
>>> -Jeff
>>> 
>> 
>> I will be recording wav format files.
>> 
>> It is my understanding that recording has to be done internally in
>> an uncompressed format so that the two legs of the
>> call can be mixed.
> 
> I'm not sure why you say "mixed" -- that would imply conferencing (i.e. more than 2 endpoints).  If you were willing
> to record in G729 format Asterisk should allow that (not sure if it does, other systems do).  Then you would not need
> transcoding, just "pass through" G729 between the call legs.  But you would need a media player that can deal with the
> compressed .wav file format when you need to play recorded data.

At least the way AMI's monitor action works there is a recording for each leg of the call, even if only two, and then soxmix is spawned to mix the two leg files into a single file. It is my understanding that in order to do this step one needs to have uncompressed sound files.

We also use chanspy to listen in on calls. This also needs something other than compressed sound as I understand it.

Maybe my whole understanding of all this is wrong but this is what I understand.

> 
>> The G729 encoding and decoding is needed between Asterisk and
>> the SIP provider due to available Internet bandwidth.
> 
> A typical requirement.
> 
>> The SIP provider talks to Asterisk via G729. The phones will be
>> configured for whatever is the best codec, maybe ulaw,
>> as they are all on the same LAN as the Asterisk box bandwidth
>> is not an issue.
> 
> Recording format seems to be the key issue -- solve that and you can configure softphones as needed.
> 
> -Jeff
> 
>>>> On Mar 22, 2010, at 5:05 PM, Rafael Prado Rocchi wrote:
>>>> 
>>>>> How many simultaneous channels?
>>>>> 
>>>>> Rafael Prado
>>>>> 
>>>>> 
>>>>>> -----Original Message-----
>>>>>> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>>>>>> bounces at lists.digium.com] On Behalf Of Jim Dickenson
>>>>>> Sent: segunda-feira, 22 de março de 2010 2:33
>>>>>> To: Asterisk User MailList
>>>>>> Subject: [asterisk-users] Transcoding question
>>>>>> 
>>>>>> We are getting ready to install a client that uses g729 when talking to
>>>>>> their SIP provider to minimize bandwidth usage. We are going to want to
>>>>>> be able to record the calls using AMI monitor actions into wav sound
>>>>>> files. All the phones are soft phone running on Windows XP systems.
>>>>>> 
>>>>>> Questions I have are what would the best codec be to have the soft
>>>>>> phone use since, as I understand it, in order to mix the audio
>>>>>> something will need to be transcoded. Can a two CPU quad core xeon 2GHz
>>>>>> system handle the transcoding load or would if be better to have a
>>>>>> daughter card handle the transcoding.
>> 

-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/