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authorKevin Harwell <kharwell@digium.com>2015-06-12 16:58:27 -0500
committerKevin Harwell <kharwell@digium.com>2015-06-15 12:35:53 -0500
commit31c77b157b84527b1a68d96f7a23c3e7b242ee99 (patch)
treec9794f599c97be7658401b977e69d1fc0e7a3bfe /CHANGES
parent2618d1e6380659441c3ab5c94a8cdd2f26c53f49 (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--CHANGES6
1 files changed, 6 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index bd79dbab7..d0363f7c3 100644
--- a/CHANGES
+++ b/CHANGES
@@ -17,6 +17,12 @@ 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.
+
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--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
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