Displaying 20 results from an estimated 9000 matches similar to: "Best Codec"
2010 Mar 17
3
SIP codec negotiation / manipulation
We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second
2011 Jun 13
5
No audio after a reinvite changing codec
Hi all,
we have a problem with a reinvite sent by our SIP provider to change audio
codec due to the recognition of a fax tone.
After that the SIP call session has been established (INVITE and 200 OK) we
have the following codec situation:
UAC ASTERISK UAS | ASTERISK UAC
PROVIDER
g711 <----------------------> g711
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)
I have 2 SIP-services to
2004 Dec 02
4
Codec Conversion
Hello,
Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything.
I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is
2004 Nov 24
2
Codec control
How can i control the codec for the calls. For example I have 3 SIP
phones registered to asterisk
The firs two are in the local area network (behind nat)- I want to use
g711 between them and to connect directly (canreinvite=yes)
and the third is in internet - want all calls to it and from it to use
g729 and media to go through asterisk.
So if Phone 1 calls Phone 2 the codec to be g711, but when
2007 Jul 03
5
Determining the used codec for the IP Trunk (SIP Trunk)
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
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2007 Apr 23
4
SIP devices with packet loss tolerance
Greetings list,
Hoping someone might have experience with poorly-performing net connections and which devices work best over them.
One of our clients has a number of employees that work from home, and are given a SIP phone to take with them and hook up to their broadband. For the most part, this works fine, but there are an increasing number where sound quality is poor ("chops" in and
2009 Oct 20
1
Is there a way to force a codec on an incoming sip uri call?
Hello,
I'd like to implement some public sip uri's that poeple can call into
and get an echo test. Is there a way to force a codec so that users
can test various codecs?
Something like:
echo-test at example.com (negotiates whatever codec, is there a way to
figure out what codec was negotiated and tell the user)
echo-test-g711 at example.com (forces g711)
echo-test-g729 at
2013 Oct 01
2
is g729 codec free? or under license???
hello all,
i have problem in using g729 codec. my asterisk version is 1.8.22. when i
run "core show codecs" in asterisk, there is a g729 codec in the list so i
assume that i can use it for my channels. but connection can not be set
when i use it for my h323 channel.
i read somewhere that codec g729 is a commercial codec and i should buy its
license in order to use it. is it true? if
2003 Aug 14
7
What is the highest quality codec I can use for recording voice messages?
I have looked at the codec's available but I don't know how get the highest
quality recorded message.
If a user calls in over the normal telephone network is this limited to the
carriers codec or the codec at the asterisk side?
Would I get a higher quality result using VoIP rather than the normal
network?
Any help would be appreciated.
Thanks
Fats.
2004 Jul 19
6
Codecs - Advantages
Hi,
I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network.
Thanks
2008 May 21
1
speex, ilbc and g729 codecs, GSM with IAX
Dears;
I do not know if any had experience in using speex or
ilbc with IAX and got good results, because I am
facing a problem with GSM.
I am facing a noise problem when I am using GSM with
IAX trunk as following:
IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk
using GSM codec ---> Remote Asterisk Box ---> Digium
Card (FXO) to terminate the call to the destination
While no
2003 Dec 16
28
codec negotiation
Hi list,
I'm with a little problem on codec negotiation between a cisco827 and
asterisk.
My sip.conf is like that:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
amaflags = default
allow=g729
allow=gsm
allow=alaw
allow=ulaw
;disallow=all
and cisco like that:
dial-peer voice 6 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:<asterisk-ip>
2006 Apr 19
1
Codec problem from SIP to H323
Hello.
I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:
- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf
2023 Jul 05
3
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
Anyone? I have hard time to believe this is not possible with chan_pjsip.
Anyway, may I ask how people handle the following scenario which I
imagine should be quite common:
- I have internal extensions talk to each other using g722. so their
codec setting (with chan_sip now) is "allow=g722,ulaw"
- I have carriers trunks that handle ulaw only (allow=ulaw)
- calls between
2006 Oct 13
1
3way calling / codec problem
I'm having problems with conference calls (3-way) when I have my codec
forced to g729 in sip.conf.
I'm using Grandstream 2000s.
If enable both g711 and g729 then 3 way calling and transfers work.
I'm not sure why this would matter?
Here's the error:
Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs!
Any help is greatly appreciated!
2007 Sep 26
2
My G729 problem re-visited
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
2005 Sep 20
6
iax2 trunking wackyness
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd, maybe there is a version
skew?
Any suggestions will be appreciated.
Thanks
Clive
2014 Jun 04
1
Renegotiate SIP audio codec after call is up
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2007 Jun 28
2
Fax passthrough howto codec upspeed
Hello everybody,
Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
The issue I have is regarding codec upspeed when the call is already connected using G729 for example. The setup is fax---ATA---asterisk---Cisco---fax
When codec upspeed should happen, ATA or Cisco will send a G711 reINVITE causing the codec to be switched over, but asterisk does NOT