[asterisk-users] Asterisk 1.4 and g723
Phil French
pfrench at caprock.com
Sat Jan 20 22:37:25 MST 2007
I appreciate the response. The ATA is the linksys SPA-2102 and some of
its configured settings are below. After the ATA information I have
included the sip.conf file and packet summary of a call with garbled
audio. Regarding the G723 codec, we have compiled a g723.1 codec. This
same source is working for Asterisk 1.2.
---start ata
settings-------------------------------------------------------
SIP Settings
SIP Transport: UDP SIP Port: 5060
SIP 100REL Enable: no EXT SIP Port:
Auth Resync-Reboot: yes SIP Proxy-Require:
SIP Remote-Party-ID: yes SIP GUID: no
SIP Debug Option: none RTP Log Intvl: 0
Restrict Source IP: no Referor Bye Delay: 4
Refer Target Bye Delay: 0 Referee Bye Delay: 0
Refer-To Target Contact: no Sticky 183: no
Auth INVITE: no
Audio Configuration
Preferred Codec: G723 Silence Supp Enable: no
Use Pref Codec Only: yes Silence Threshold:
medium
G729a Enable: yes Echo Canc Enable: yes
G723 Enable: yes Echo Canc Adapt Enable: yes
G726-16 Enable: no Echo Supp Enable: yes
G726-24 Enable: no FAX CED Detect Enable: yes
G726-32 Enable: no FAX CNG Detect Enable: yes
G726-40 Enable: yes FAX Passthru Codec: G711u
DTMF Process INFO: yes FAX Codec Symmetric: yes
DTMF Process AVT: yes FAX Passthru Method: None
DTMF Tx Method: InBand DTMF Tx Mode:
Strict
FAX Process NSE: yes Hook Flash Tx Method: None
FAX Disable ECAN: yes Release Unused Codec: yes
FAX Enable T38: yes FAX T38 Redundancy: 0
FAX Tone Detect Mode: caller or callee
---end ata
settings-------------------------------------------------------
---start sip.conf-------------------------------------------------------
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
minexpiry=60
t1min=1000
t38pt_udptl = yes
canreinvite=yes
[authentication]
[pstn_gw1]
type=friend
host=172.17.17.20
disallow=all
allow=g723
context=us
qualify=500
insecure=port
[sip_ata_01]
type=friend
secret=password
host=dynamic
disallow=all
allow=g723
context=us
dtmfmode=RFC2833
qualify=2000
[sip_ata_02]
type=friend
secret=password
host=dynamic
disallow=all
allow=g723
context=us
dtmfmode=RFC2833
qualify=2000
---end sip.conf---------------------------------------------------------
---start packet summary-------------------------------------------------
SRC DST Protocol Info
ATA AST SIP/SDP Request: INVITE sip:2000 at ASTERISK, with session
desc.
AST ATA SIP Status: 407 Proxy Authentication Required
ATA AST SIP Request: ACK sip:2000 at ASTERISK
ATA AST SIP/SDP Request: INVITE sip:2000 at ASTERISK, with session
desc.
AST ATA SIP Status: 100 Trying
AST ATA SIP/SDP Status: 200 OK, with session description
ATA AST SIP Request: ACK sip:2000 at ASTERISK:5060
ATA AST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169,
Seq=56662, AST ATA G.723.1 Payload type=ITU-T G.723,
SSRC=1606418004, Seq=59141, AST ATA G.723.1 Payload type=ITU-T
G.723, SSRC=1606418004, Seq=59142, ATA AST G.723.1 Payload
type=ITU-T G.723, SSRC=2925445169, Seq=56663, AST ATA G.723.1
Payload type=ITU-T G.723, SSRC=1606418004, Seq=59143, ATA AST
G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56664, AST
ATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59144, ATA
AST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56665, ATA
AST G.723.1 Payload type=ITU-T G.723, SSRC=2925445169, Seq=56666, AST
ATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004, Seq=59145,
#truncated for the sake of brevity#
ATA AST SIP Request: BYE sip:2000 at ASTERISK:5060
AST ATA G.723.1 Payload type=ITU-T G.723, SSRC=1606418004,
Seq=59174, AST ATA G.723.1 Payload type=ITU-T G.723,
SSRC=1606418004, Seq=59175, AST ATA G.723.1 Payload type=ITU-T
G.723, SSRC=1606418004, Seq=59176,
AST ATA SIP Status: 200 O
---end packet summary---------------------------------------------------
Thanks,
Phil
Phil French
Systems Engineer
-------------------------------
CapRock Communications
4400 S. Sam Houston Parkway E.
Houston, Texas 77048
Office: 832 668 2643
pfrench at caprock.com
www.caprock.com
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-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew
Joakimsen
Sent: Saturday, January 20, 2007 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 and g723
What G723 codec do you have on Asterisk? What is your SIP.CONF? What
ATA/Phone is being used and what are the exact settings, especially
for the codec?
On 1/19/07, Phil French <pfrench at caprock.com> wrote:
> I am setting up Asterisk for use in a low bandwidth environment. As
> bandwidth is precious and our ATA's support it, the decision was made
to
> use the g723 codec. I have been working on this for a few days and
have
> not been successful. The issue that I am having is garbled noise at
the
> client on calls whose RTP streams are terminated by Asterisk system.
> This is the case for all the dialplan applications I have tested
except
> for Echo. The critical application for us is Voicemail. When a call
to
> voicemail extension is initiated the Asterisk console does not
indicate
> any error. Packet captures indicate the call is active and streaming
> g723 data. Everything seems well but is not. The audio at the client
> is unrecognizable. One thing that I have noticed is that the bitrates
> in the upstream and downstream direction differ. From Asterisk to ATA
> the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s. From ATA to
> Asterisk the bitrate is a constant 6.3 kb/s. I don't think this is a
> problem but seems odd. As a comparison I captured packets from a call
> to the echo application and found that the bitrate was 6.3 kb/s in
both
> upstream and downstream packets. Additionally, all prompts are g723
> format. Voicemail is saved as g723sf. As a parrallel task a
co-worker
> has gotten 1.2 to work with g723. However we require 1.4 for t.38
> pass-through.
>
> The end-to-end system is illustrated below.
>
> [Asterisk]
> / \
> ip ip
> / \
> [PSTN]--pri--[GATEWAY]--ip--[ATA]<--2pr-->[Phone]
>
> System details
> -Asterisk server version 1.4 - compiled from source - Fedora Core 6
> -Gateway - Cisco 2811 -ATA - Linksys 2102
>
> I would appreciate any advice or suggestions. It should be noted that
> the calls to the PSTN through the gateway and calls between ATA's are
> working fine.
>
> Regards,
>
> Phil French
>
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