G.726 is an ITU standard codec. This codec uses the Adaptive Differential Pulse Code Modulation (ADPCM) scheme.
Like G.711, G.726 has its roots in the PSTN network. It is primarily used for international trunks to save bandwidth. Where G.711 uses 64 Kbps, G.726 uses 32 Kbps, providing nearly the same quality. It is also the standard codec used in DECT wireless phones.
The bitrate can be 16, 24, 32 or 40 Kbps, but 32 Kbps is the defacto standard.
Asterisk currently supports the 32kbps standard only.
In addition to the one that comes with asterisk, one asterisk compatible exists in spandsp version 0.0.2pre26 and later. By some rumors this should be closer to the standards specification
than the one from digium
ITU G series reccomendations
Like G.711, G.726 has its roots in the PSTN network. It is primarily used for international trunks to save bandwidth. Where G.711 uses 64 Kbps, G.726 uses 32 Kbps, providing nearly the same quality. It is also the standard codec used in DECT wireless phones.
The bitrate can be 16, 24, 32 or 40 Kbps, but 32 Kbps is the defacto standard.
Asterisk currently supports the 32kbps standard only.
In addition to the one that comes with asterisk, one asterisk compatible exists in spandsp version 0.0.2pre26 and later. By some rumors this should be closer to the standards specification
than the one from digium
ITU G series reccomendations
Comments
333Asterisk g.726 CISCO PAP2NA ATA related problem
I am quite a newbie to asterisk, mine is a non commercial strictly academic experimental asterisk setup consisting of a Asterisk Server mounted on a Redhat EL3 with Cisco PAP2 NA ATA's being also connected to the backbone (ethernet) so as to make calls between two analogue phones.
To start with I used g.711 codec and everything worked fine, but cause it is a bandwidth "exhaustive" codec i have to now resort to some other codec. As my implementation is academic in nature so I do not wish to purchase the license for g729 right now and wud prefer working on something that is freely downloadable. The following codecs r not spported by the PAP2 NA
gsm, speex, ilbc
What is supported is g.726, though in my case it works only with 32 kbps (that also partially, read on), that is i permit 32 kbps bandwidth from my pap2 after declaring the same alllow=g726 in my sip.conf file.
My Problem is even with g.726 I am straightway going into voicemail instead of a direct call between the two phones (analogue phones connected to the ATA), next problem is I do not know from my asterisk end what bandwidth am i configuring when i mention allow=g726 in my sip.conf. Next I wish to use g726 in its 16 kbps mode, can someone help me out as to how actually can it be done, furthermore if anyone can suggest a way to use anyother codec other than g.711 with CISCO PAP2 NA ATA's with BANDWIDTH CONSUMPTION NOT more than 16 kbps, I wud be eternally thankfull.
(and no i dont want to use g.723 cause it doesn't support voicemail as per asterisk docs!!!)
Any links references or archived emails in this regard wud be welcome.
Regards
A Desperate being
Xeron