Hi all, Iâ€™m new to Asterisk, thus far I have spent a little time playing with a system installed by somebody else and Iâ€™ve quickly read through the excellent Asterisk book.
Asterisk is obviously aimed squarely at telecommunications which it seems to do very well but Iâ€™m wondering how easy/possible it would be to modify the system for my application.
I am looking at ways of transmitting audio feeds over IP networks, mostly over LAN or dedicated Ethernet connections but also over the wider internet as a whole, in a very reliable and minimum latency fashion.
I need to have support for the following codecs:
MPEG 1 layer 2 (rates from 32kHz mono (56kbps) through to 48kHz stereo (384kbps))
MPEG 2 layer 2 (rates from 16kHz mono (32kbps) through to 48kHz stereo (384kbps))
Uncompressed 12, 16, 20 and 24 bit PCM (no companding) at rates of 32kHz and 48kHz
The above are essential but I would also like to provide higher sampling rate PCM up to 192kHz 24bit, MP3 (various bitrates) MPEG-4 AAC (various bitrates) and Dolby AC3 up to 640kbps.
Others may be added to the list.
I know Asterisk natively supports some of these but could it easily support more? Can it handle the much higher bitrates?
What would a typical round trip time be over a LAN using uncompressed PCM codecs that require very little CPU time to packetise the stream?
Does Asterisk provide any features for Session Announcement Protocol (SAPv1 (RFC2974)), Session Description Protocol (SDP (RFC4566)) and Real Time Control Protocol (RTCP)?
Iâ€™m also looking at the PJSIP toolkit (pjsip.org/) as a candidate for developing this system. Does anybody have any experience of PJSIP Vs Asterisk? Which do you think might be more suitable for my application?
Thanks very much in advance for any thoughts.