[asterisk-users] My G729 problem re-visited

[Power, Paul C.]

Try the Prescott version of the G729 .so.
That one is made for xeon's.


________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike
Lynchfield
	Sent: Friday, October 12, 2007 2:36 PM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: Re: [asterisk-users] My G729 problem re-visited
	
	
	How do you get 11ms translation time on ulaw 729 ?
	
	we have 12ms and its dual xeons 2.6..
	
	
	On 9/26/07, Scott Moseman < scmoseman at gmail.com
<mailto:scmoseman at gmail.com> > wrote: 

		Ok, I built a test system to duplicate my problem and
provide myself 
		a platform that I can mess around with to try and break
any features.
		My problem is G729 pass-through from a gateway to a
phone. I think
		I even have transcoding working, which makes me more
confused on
		what's wrong with my pass-through. It must be a
configuration issue. 
		
		The basics...
		
		*CLI> core show version
		Asterisk 1.4.11 built by root @ fwd-tst02 on a i686
running Linux
		
		*CLI> show modules like 723
		Module Description Use Count
		codec_g723.so G.723.1 Coder/Decoder 0 
		format_g723.so G.723.1 Simple Timestamp File Format 0
		
		*CLI> show modules like 729
		Module Description Use Count
		codec_g729.so G.729 Coder/Decoder 0
		format_g729.so Raw G729 data 0
		
		*CLI> show translation 
		[truncated]
		g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex
ilbc g726 g722
		ulaw 5 2 - 1 2 2 1 3 7 - 11 2 -
		alaw 5 2 1 - 2 2 1 3 7 - 11 2 -
		g729 5 2 2 2 2 2 1 3 - - 11 2 -
		
		The configuration... 
		
		[gateway]
		type=friend
		host=gateway
		context=default-inbound
		disallow=all
		allow=g729
		
		[phone]
		type=friend
		context=sip
		host=dynamic
		username=phone
		secret=scott
		dtmfmode=RFC2833
		disallow=all
		allow=g729
		callerid=Scott
		qualify=yes
		canreinvite=no
		
		exten => 1266,1,Dial(SIP/[number],30,t)
		exten => 1266,2,Congestion
		
		exten => 1266,1,Dial(SIP/[number],30)
		exten => 1266,2,Congestion 
		
		(The same results using both of the above dialplans...)
		
		The environment...
		
		PSTN -> Gateway -> Asterisk -> Phone
		
		What I'm seeing works...
		
		With the gateway setup to send both G711 and G729, it
sends 
		an INVITE which includes both G711 and G729 codecs.
Asterisk
		sends an INVITE to my phone with only G729. The call is
made
		and there's a conversation in G711 with the gateway and
G729
		with the phone. I assume this means Asterisk is
transcoding. 
		
		What I"m seeing fails...
		
		With the gateway setup to send only G729, it sends an
INVITE
		to Asterisk which includes only G729. Asterisk send an
INVITE
		to the phone using G729, too. The 200 OK from the phone
to 
		the Asterisk includes G729. The 200 OK going from
Asterisk to
		the gateway doesn't include ANY codec. The call is
dropped the
		moment I pickup the phone to answer the call.
		
		My question...
		
		Why does Asterisk not want to respond to my gateway in
G729? 
		Even if the gateway requests it, Asterisk seems to just
ignore it.
		From the transcoding call, and phone to phone G729
calls, I have
		proof that Asterisk knows how to handle G729 calls.
		
		Where do I go from here??? 
		
		Thanks,
		Scott
		
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