[Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 ,
fedora 1 , asterisk 1.0.6
Bashir Ullah
Bashir.Ullah at lamsre.com
Thu Jul 28 03:08:09 MST 2005
Hi Apu
Thanks for reply,
i was using same config with PIII, and did not face any outgoing voice
delay. after i shift with xeon then i am geting this trable. and in that
time i was using g729a.so for i586, and now i am changing this to i686, just
this i change nothing else. same setup same getway , same endpoint. i dont
understand why this happening.
here for any suggession i attach my "show translation " pic and my
oh323.conf setup.
please let me know am i missing anything or need to change or add anything.
Thanks.
Bashir Ullah
iax*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - 2 2 2 2 2 1 5
10 - 16
gsm 11 - 2 2 2 2 1 5
10 - 16
ulaw 11 2 - 1 2 2 1 5
10 - 16
alaw 11 2 1 - 2 2 1 5
10 - 16
g726 11 2 2 2 - 2 1 5
10 - 16
adpcm 11 2 2 2 2 - 1 5
- 16
slin 10 1 1 1 1 1 - 4
9 - 15
lpc10 12 3 3 3 3 3 2 -
11 - 17
g729 12 3 3 3 3 3 2
- - 17
- - - - - - - - -
- -
ilbc 12 3 3 3 3 3 2 6
11 - -
and my oh323.conf
; Configuration file of OpenH323 channel driver
;-----------------------------------------
; General configuration options (ports, jitter, GK, ...)
;-----------------------------------------
[general]
; Address to bind to for incoming connections. Default is ALL.
listenAddress=0.0.0.0
; Port to listen to. Default value is 1720.
listenPort=1720
; Port to connect to. (Used only when we don't have a gatekeeper) Default
value is 1720.
connectPort=1720
;Configure TCP port range to be used by H.323
tcpStart=10000
tcpEnd=20000
;Configure UDP port range to be used by H.323
;Note: The port range used by RTP are configured from
;"rtp.conf"
udpStart=10000
udpEnd=20000
;Enable fast start (yes,no).
fastStart=no
;Enable H.245 tunnelling (yes,no).
h245Tunnelling=no
;Enable early H.245 messages in call SETUP message.
h245inSetup=no
;Enable in-band-DTMF detection. (Note: Netmeeting uses in-band DTMFs)
inBandDTMF=yes
;Enable silence suppression.
silenceSuppression=no
;Set jitter buffer (in milliseconds, 20...10000).
jitterMin=20
jitterMax=10000
;Set IP Type-of-Service byte for RTP channels. Valid values for this option
are:
;lowdelay, throughput, reliability, mincost, none
ipTos=lowdelay
;Set the maximum number of inbound/outbound/simultaneous H.323 connections.
outboundMax=20
inboundMax=20
simultaneousMax=20
;Set the bandwidth limit for H.323 connections. The value is in Kbps.
;bandwidthLimit=2048
;Set tracing options for the wrapper library and for the OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
; "crlCallNumber", "crlCallTime", "crlThreshold" - Call Rate Limiter params
; (ingress direction). When the total number of active calls is above
; 'crlThreshold' then the rate of the incoming H.323 calls is restricted
; in a way where no more than 'crlCallNumber' calls are allowed in
; 'crlCallTime' milliseconds, thus limiting the rate of incoming calls
to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;crlCallNumber=
;crlCallTime=
;crlThreshold=
; "language", "musiconhold" - The default language and music-on-hold class
; for the OH323 channels.
;language
;musiconhold
;Disable gatekeeper or specify a gatekeeper. Valid values for this option
are:
;DISABLE,
;DISCOVER,
;<gatekeeper's DNS name>,
;<gatekeeper's ip>,
;GKID:<gatekeeper's id>
gatekeeper=DISABLE
;gatekeeper=65.61.200.138
;Set the gatekeeper password
;gatekeeperPassword=secret
;Set the gatekeeper registration timeout
;gatekeeperTTL=600
;Set the mode for sending user-input Valid values for this option are:
;Q931 - Q.931 Keypad Information Element
;STRING - H.245 string
;TONE - H.245 tone
;RFC2833 - RFC2833
userInputMode=TONE
;AMA flags (default, omit, billing, documentation)
amaFlags=default
;Account code
accountCode=H323
;Set the default context of H.323 calls.
context=H323
;-----------------------------------------
;Configure H.323 aliases, prefixes and
;related ASTERISK's contexts
;-----------------------------------------
[register]
;Aliases/prefixes associated with the default context defined in section
[general].
alias=asterisk
alias=123
;Aliases/prefixes routed in "all-aliases" context.
context=all-aliases
alias=ASTERISK
alias=666
;Aliases/prefixes routed in "more-aliases" context.
context=more-aliases
alias=665
;Aliases/prefixes routed in "more-aliases" context.
context=sip
alias=bashir
;Aliases/prefixes routed in "all-prefixes" context.
;context=all-prefixes
;gwprefix=00
;gwprefix=01
; Aliases/prefixes routed in "more-stuff" context.
;context=more-stuff
;alias=664
;gwprefix=02
;Aliases/prefixes routed in "netmeeting " context.
context=oh323
;gwprefix=73596
;gwprefix=99123
gwprefix=989
gwprefix=01188031
;-----------------------------------------
;Specify and configure CODEC related options
;-----------------------------------------
[codecs]
;Define the codec list of the channel driver. Every "codec" option may have
a "frames" option
;associated with it. Valid values for the "codec" option are:
; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72315K3 - G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G726 - G.726(32k)
; G72616K - G.726(16k)
; G72624K - G.726(24k)
; G72632K - G.726(32k)
; G72640K - G.726(40k)
; G728 - G.728
; G729 - G.729
; G729A - G.729A
; G729B - G.729B
; G729AB - G.729AB
; GSM0610 - GSM 0610
; MSGSM - Microsoft GSM Audio Capability
; LPC10 - LPC-10
;
;G.729
;codec=G729
;frames=2
;G.729A
;codec=G729A
;frames=2
;G.729B
;codec=G729B
;frames=2
;G.729AB
;codec=G729AB
;frames=2
;G.723.1(5.3k)
;codec=G72315K3
;frames=2
;G.723.1(6.3k) used by mera,
;codec=G7231
;frames=0
;G.723.1(6.3k)
;codec=G72316K3
;frames=2
;G.723.1A(6.3k) used by quintunm,
;codec=G7231A6K3
;frames=2
;G.711 u-Law
;codec=G711U
;frames=20
;G.711 A-Law
;codec=G711A
;frames=20
;G.726(32k)
;codec=G726
;frames=20
;G.726(16k)
;codec=G72616K
;frames=20
;G.726(24k)
;codec=G72624K
;frames=20
;G.726(32k)
;codec=G72632K
;frames=20
;G.726(40k)
;codec=G72640K
;frames=20
;G.728
;codec=G728
;frames=20
;G.729
;codec=G729
;frames=0
;G.729A
;codec=G729A
;frames=2
;G.729B
;codec=G729B
;frames=2
;G.729AB
;codec=G729AB
;frames=2
;GSM 0610
;codec=GSM0610
;frames=4
;Microsoft GSM Audio Capability
;codec=MSGSM
;frames=4
;LPC-10
;codec=LPC10
;frames=20
;G.729
;codec=G729A
;frames=24
codec=G729A
frames=24
;G.723.1A(6.3k) used by quintunm
codec=G72316K3
frames=24
----- Original Message -----
From: "Apu Islam" <apuislam at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, July 28, 2005 12:14 AM
Subject: Re: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8
,fedora 1 , asterisk 1.0.6
several factors :
- check 'show translation' from asterisk to see how long it will take
for transcoding between your codecs. with your machine, should not be
long.
- the h323 endpoints latency. ( a lot of times this attributes to delay.)
- echo cancellation and zitter buffer ( zitter significantly improves this.)
-apu
On 7/27/05, Bashir Ullah <Bashir.Ullah at lamsre.com> wrote:
>
> Hi All
>
> I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2
gb
> ram, with g729 for i686 , (fedora 1).
>
> my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
> otherparty realtime voice , but other party geting sip party's voice 1 sec
> later (not realtime).
>
> please some help me to solve this issu, last one month i am tring
different
> different way to solve this issu.
>
> is it codec problem or something else.
>
>
> thanks
>
> bashir
> _______________________________________________
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