diff options
| author | Kevin Harwell <kharwell@digium.com> | 2015-06-12 16:58:27 -0500 |
|---|---|---|
| committer | Kevin Harwell <kharwell@digium.com> | 2015-06-15 12:40:03 -0500 |
| commit | 93ac45d3bd89580776cb388f288861ec3545d7a7 (patch) | |
| tree | 63228dfe934e3812804bb5b2911635d97128e840 /CHANGES | |
| parent | b8bc15286fd4610221e98f53c34ab486f357198e (diff) | |
res_pjsip: Add option to force G.726 to be treated as AAL2 packed.
Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 6 |
1 files changed, 6 insertions, 0 deletions
@@ -186,6 +186,12 @@ AMI * A new ContactStatus event has been added that reflects res_pjsip contact lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown. +res_pjsip +------------------ +* A new 'g726_non_standard' endpoint option has been added that, when set to + 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it + is AAL2 packed on the channel. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------ ------------------------------------------------------------------------------ |