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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-03-17 21:30:19 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2005-03-17 21:30:19 +0000
commit235a6486c3ca01b9bdac577774cc90fa4f06a58f (patch)
tree2ff53ca559c15101655e44ad4281225c8aae6685 /configs
parent98aa637d27b5ac36a35fc85b596f00d1104398e3 (diff)
Add PLC and jitter buffer and iax2 meta trunk with timestamps (bug #2532, #3400)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5192 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rwxr-xr-xconfigs/codecs.conf.sample6
-rwxr-xr-xconfigs/iax.conf.sample27
2 files changed, 33 insertions, 0 deletions
diff --git a/configs/codecs.conf.sample b/configs/codecs.conf.sample
index a6d7ea16a..ec3c496e3 100755
--- a/configs/codecs.conf.sample
+++ b/configs/codecs.conf.sample
@@ -15,3 +15,9 @@ abr => 0
vbr_quality => 5
; true / false
dtx => false
+
+[plc]
+; for all codecs which do not support native PLC
+; this determines whether to perform generic PLC
+; there is a minor performance penalty for this
+genericplc => true
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 6da635789..b0ed7e1e7 100755
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -61,6 +61,12 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; The jitter buffer's function is to compensate for varying
; network delay.
;
+; There are presently two jitterbuffer implementations available for * and chan_iax2;
+; the classic and the new, channel/application independent implementation. These
+; are controlled at compile-time. The new jitterbuffer additionally has support for PLC
+; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost
+; packets.
+;
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
; will be dejittered by the jitter buffer at the other end.
@@ -68,9 +74,16 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; jitterbuffer=yes|no: global default as to whether you want
; the jitter buffer at all.
;
+; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
+; we don't want to do jitterbuffering on the switch, since the endpoints
+; can each handle this. However, some endpoints may have poor jitterbuffers
+; themselves, so this option will force * to always jitterbuffer, even in this case.
+; [This option presently applies only to the new jitterbuffer implementation]
+;
; dropcount: the jitter buffer is sized such that no more than "dropcount"
; frames would have been "too late" over the last 2 seconds.
; Set to a small number. "3" represents 1.5% of frames dropped
+; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; maxjitterbuffer: a maximum size for the jitter buffer.
; Setting a reasonable maximum here will prevent the call delay
@@ -81,18 +94,22 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; the jitter buffer can end up bigger than necessary. If it ends up
; more than "maxexcessbuffer" bigger than needed, Asterisk will start
; gradually decreasing the amount of jitter buffering.
+; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; minexcessbuffer: Sets a desired mimimum amount of headroom in
; the jitter buffer. If Asterisk has less headroom than this, then
; it will start gradually increasing the amount of jitter buffering.
+; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
;
; jittershrinkrate: when the jitter buffer is being gradually shrunk
; (or enlarged), how many millisecs shall we take off per 20ms frame
; received? Use a small number, or you will be able to hear it
; changing. An example: if you set this to 2, then the jitter buffer
; size will change by 100 millisecs per second.
+; [This option is not applicable to, and ignored by the new jitterbuffer implementation]
jitterbuffer=no
+forcejitterbuffer=no
;dropcount=2
;maxjitterbuffer=500
;maxexcessbuffer=80
@@ -100,6 +117,16 @@ jitterbuffer=no
;jittershrinkrate=1
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
+
+; Should we send timestamps for the individual sub-frames within trunk frames?
+; There is a small bandwidth use for these (less than 1kbps/call), but they ensure
+; that frame timestamps get sent end-to-end properly. If both ends of all your trunks
+; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you
+; can probably suppress these. The receiver must also support this feature, although
+; they do not also need to have it enabled.
+;
+; trunktimestamps=yes
+
;
;
; We can register with another IAX server to let him know where we are