How can i use G729 with JSSIP?
I am developing an app with asterisk integration with WRTC, but in call asterisk report this error: res_pjsip_sdp_rtp.c:418 set_caps: No joint capabilities for ‘audio’ media stream between our configuration((g729)) and incoming SDP((ulaw|alaw|g722|opus)).
however my equips dont suport this codecs ( (ulaw|alaw|g722|opus ), how can i solved it !
Unless things have changed the browser does not do G729.
and the Codec g711, where i found for download ?
To which side are you referring to? The browser and Asterisk both support ulaw. You just have to configure it in your pjsip.conf endpoint to be allowed.
I am reading about transcoding g711 to g729, because my equips dont work in both codec.
How to make this transcoding ?
or don’t exist the possibility?
Install DIgium’s G.729 codec and pay for and install the licences.
Please note that I have never heard of a SIP device that does not support G.711.
I’m using an ATA for analog phones and an IP phone
they contain the codecs g729, g711u, g711a, iLBC, g723 and g726
However the system I am developing is via browser and contains the codecs ulaw, alaw and opus
how can I get around this situation ??
g711u = ulaw (G.711 µ Law - North America and Japan)
g711a = alaw (G.711 A Law - Rest of the World))
just one more question, in my pjsip in the allow field should i put as alaw / ulaw or g711u / g711a ??
ulaw or alaw depending on the environment in which your system operates. Using the wrong one will result in a slight degradation when the media is exchanged with the PSTN, assuming that the PSTN part of the call only uses one of them.