About SIP CONFIG & PHPAGI

hi,

Will the asterisk can adjust the recording volume loud it?
Also, if using a SIP to connected, like server A use sip to connect the server B, Can I use echo cancellation? and have a way to transfer large call voice?
Like CHAN_DAHDI.CONF
echocancler = yes (open Echo Cancellation)
txgain = 4 (increased transmit voice)
rxgain = 4 (increased receive voice)

AND,

In dialplant use phpagi recording,
$agi->exec (‘MixMonitor’,’/var/voice/{$caller}.wav’);
I know the MIXMONITOR command can use parameter conversion in the recording has been completed, but this is not what I want
Excuse me whether by certain commands let MixMontitor recording started with MP3 type recording?
((if it cannot use the MP3 type, will it have some type to be select))

thanks

You shouldn’t need echo cancellation on SIP because the phone should do it.

I believe that the audiohook mechanism in later versions of Asterisk can adjust the audio volume, although this is better done in the originating phone, before the data has been quantised to digital form. Once the data is in digital form, there will be some information loss if you try and change its gain, and, if the sound is too loud, it will have been clipped at the point at which it became digital, so that cannot be undone.

Recording in MP3 doesn’t make a lot of sense for telephone calls, as the intrinsic quality and sample rates are much lower than those of MP3. There may also be issues still with patents on MP3 encoding.

Thank you very much for your answer,
I know the sound quality and file size is proportional,
However, if the transmission by e-mail, etc., then I think the file is really too big,
MONITOR can set the audio file type, but MIXMONITOR whether there is an option to set?

I also know the size of the sound in the user terminal adjustment is relatively good, but some brands of telephones are relatively quietly , if it is adjusted by ASTERISK Well, after all, because isn’t replace one or two things, there are dozens of questions.

at sip channel, if i change the codes to G.711 , is it voice will bettered than G.729?

But in any case, thank you for your answer again,
Thank you david55!

G.729 will produce smaller files, but worse audio. G.711 is PSTN telephone quality. Whilst there are some codecs that produce better quality than G.711, they are only used for internal calls, and mainly for executive phones. (The advantage of G.729 is that it produces small files with less degradation in speech quality than alternative low bit rate codecs.)

See en.wikipedia.org/wiki/Mean_opinion_score for the subjective quality of various codecs. G.711 has perceptible, but not annoying, impairment, but G.729a has slightly annoying impairment.

If the receiving phone is not loud enough, you need to replace the phone, as telephone audio has limited dynamic range and the sender should already be using most of that range. If you try to amplify, you will simply clip the audio, causing distortion.

Why don’t you record a .wav file and have a script that converts the .wav file to .mp3 and deletes the .wav file (if you need to save storage) after the recording is done. I think this is a much better way to tackle this problem.

Or use WAV, rather than wav, which will create a GSM coded file, which is a sensible format for telephony speech.