Displaying 20 results from an estimated 3000 matches similar to: "Transcoding OPUS?"
2013 May 18
1
Opus in VOIP
Hi!
I'd like to ask whether someone did test Opus in real-world VOIP (SIP). Did
someone e.g. some characterization about sending faxes or DTFM through
Opus? Does it work and if yes for which bitrates?
Thanks!
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2013 Dec 15
3
Why doesn't Asterisk try to prevent transcoding
Let's say I have two devices configured and the follow call scenarios occur.
[100]
disallow=all
allow=g722&ulaw
Polycom phone with g722,ulaw,alaw,g729
[101]
disallow=all
allow=ulaw
Polycom phone with g722,ulaw,alaw,g729
101 dials 100 -> ulaw to ulaw is chosen
100 dials 101 -> g722 to ulaw is chosen
Ideally when 100 dials 101 ulaw would be chosen since it is the common
format.
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
2014 May 12
1
new install: no re-invite and unwanted transcoding
I am unable to get re-invite to work on a new system. Also, unwanted
transcoding is occurring on PSTN calls.
The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will
eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2,
CentOS 5.8) currently in production. Both systems are on VPS with public
IP addresses. Goals for the new system include: HD (g722) connections on
2011 Apr 21
1
Transcode ulaw/g722 problem
We are getting the following on 1.8.3 and 1.8.4-rc2, HELP!
Why is Asterisk unable to transcode to/from ulaw and g722?
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw, while native formats is 0x1000 (g722) read/write = 0x1000 (g722)/0x1000 (g722)
[2011-04-21 09:51:34] WARNING[22067]: chan_sip.c:6064 sip_write: Asked to transmit frame type ulaw,
2013 Aug 22
1
Strange ogginfo result
Hi!
I'm using the opus-tools package from the current Debian (it says it is
0.1.2). The output of opusinfo is strange:
$ opusinfo demo01.opus
Processing file "demo01.opus"...
New logical stream (#1, serial: 0000456a): type opus
Encoded with libopus 1.0.1-rc3
User comments section follows...
ENCODER=opusenc from opus-tools 0.1.5
Opus stream 1:
Pre-skip: 356
Playback
2009 Oct 13
3
strange transcoding values
Hello guys,
i have a question about a voip gateway we use.
I saw those values typing in cli:
core show translation
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 slin16
g723 - - - - - - - - - - - - - -
gsm - - 2001 2001 6000 2001 2000 16000 - 34002 - 6000
2007 Jan 11
1
Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!
Hi all,
I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly
i cannot dial extensions 4XXX from SIP Phones.
Now comes the wired stuff... I can dial this extensions from IAX phones as
well as from Analogue extensions connected to our legacy pbx, that is
installed on front of asterisk.
So :
Zapata Calls to SIP extensions 4XXX - OK
IAX to SIP 4XXX-OK
SIP to SIP 4XXX -
2019 Jul 08
3
opus codec
Hi All,
I am trying to get the opus codec working with linphone.
I followed the instructions... This shows me its loaded
core show translation paths opus
--- Translation paths SRC Codec "opus" sample rate 48000 ---
opus:48000 To g723:8000 : No Translation Path
opus:48000 To ulaw:8000 : (opus at 48000)->(slin at 48000
)->(slin at
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
After I have re-read the "PJSIP Advanced Codec negotiation" document, it
occurred to me that the desired behavior should actually happen
automatically, just due to the codec negotiation logic, but it looks
like asterisk doesn't actually follow the described logic which is
likely a bug.
Can you please follow with me through a simple sip call and see if I'm
missing
2011 May 04
1
asterisk 1.4.35 to 1.4.41
Under 1.4.35 I get this message printed MANY times
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000 (g722)(4096)
[May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame
type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000
(g722)(4096)/0x1000
2017 May 12
3
pjsip: asterisk can't decide which codec to use
Hello!
I'm facing completely choppy sound. The wireshark trace shows, that
there are a lot of codec changes without any trigger (means no options
or reinvite or any other package).
Background:
The call is initiated by asterisk and is received by the same asterisk
conference room via
Phone extension -> asterisk -> provider A -> provider B -> asterisk.
Asterisk initially sends
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2006 Nov 04
1
Pass through
Hi!
I want to tell asterisk to simply pass-through any codecs that my phones
support. I have to use codecs that are not popular and implemented by a
third-party, asterisk has nothing to do with them.
I've made a test with g722 (that asterisk doesn't support), i've set all my
two snom 300 phones to support only g722 and asterisk declined the sip
invitation. That is bad for me. Is it
2023 Jul 05
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello Michael,
you are referring to the following behavior - did I get it correctly?:
outbound broken: asterisk offers g722 / g711 to provider (callee),
callee answers g711. Asterisk now transcodes between caller and callee
(g722 <-> g711).
inbound works: call from provider: g711 -> asterisk drops g722 and
passes g711 to internal callee -> no transcoding.
As far as I know,
2008 Feb 07
6
Asterisk G722
Hello,
I have some problems to use G722, when my client sent an invite request
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.
I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??
Thanks.
Rachid.
Below wireshak trace:
2010 Jun 26
2
Codec negotiation
I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my voip provider and transcoding these to ulaw. In
sip.conf for the provider I have deny=all and allow=ulaw. This can
cause potential audio degrading and wastes cpu cycles.
2018 Jun 16
2
Only 8kHz recorded after disallowing all but G722 codec on inbound
We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.
My test calls show inbound to the proxy is recorded at 16kHz, inbound in
Asterisk is only 8kHz, and the peers receive 8kHz. So
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2009 Dec 07
1
g722 question
Hello,
I am working with several SIP projects that use g722, or are trying to
do so, with pjsip library.
According to pjsip team's interpretation of g722, it works with 14bits
PCM for input/output, so pjsip basically 'converts' the audio sample
from 16 bits to 14 when encoding and vice-versa. Some implementations
don't do 16<->14 bits conversion, so when pjmedia talks to