Displaying 20 results from an estimated 3000 matches similar to: "Are the Siren7 and Siren14 the G.722 HD voice codecs?"
2009 Apr 21
4
Polycom wideband codecs?
Doing a little research before Friday's Voip Users Conference call with
Dan Behringer.
Are any of the newer Polycom wideband codecs implemented in v1.6?
Specifically, G.722.1 or G.722.2?
Thanks,
Michael Graves
mgraves <at> mstvp.com
o(713) 861-4005
c(713) 201-1262
sip:mjgraves at mstvp.onsip.com
skype mjgraves
2012 Feb 09
2
Help with Codes and Polycom Phones
Hi All,
This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these:
G722
Siren14.24kbps Siren22.32kbps
Siren14.32kbps Siren22.48kbps
Siren14.48kbps Siren22.64kbps
G7221.16kbps
2015 Aug 10
2
Siren7 for Asterisk 13.5
> A Siren codec is not currently available and the one for 12 will not
> work. I have no timeframe for when this might change.
So the only option is to build one from the Polycom sources? I'm
already doing this for Siren14 (I forget why).
2009 Oct 23
3
SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk
and I'm trying to get it to accept a SIREN14 call from Polycom's softphone.
Having trouble with SDP negotiation, I want to only allow SIREN14 and
nothing else. I also want to record and playback files, any tips on what
the Record function parameters should be?
In sip.conf I have:
disallow=all
2015 Aug 07
2
Siren7 for Asterisk 13.5
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0?
Since there's nothing later, does the version for 12.0 work?
2011 Jan 05
3
VoIP PoE phones for restaurant (kitchen)
On Tue, 4 Jan 2011, Andy Graybeal wrote:
>> The Polycom 321 has not been EOL'd and supports VLAN. It is, however,
>> lacking a 2nd ethernet port if you were to go that route.
>>
>> -M
>>
> Thanks for the response Mark. I see the 331 has two ports and the same
> features as the 321.
>
> I'm wondering what phone would be best being used as an
2013 Mar 15
2
Disagreements between codec_siren14 and Polycom sources
There appears to be a disagreement between the encoding given in the
sources for Siren14 that are downloaded from Polycom (and the ITU, both
are the same) and that implemented by codec_siren14.so. The latter
agrees with the actual device.
If I make a .sln32 file and run the encoder from ITU/Polycom with
encode 0 foo.sln32 foo.siren14 48000 14000
the resulting file doesn't play back
2013 Feb 28
1
Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
seems odd that there's no transcoder directly to slin32 since anything
else will lower fidelity. But, more importantly, there is transcoding
from siren14 to slin16 and
2009 Sep 24
2
Digium transcoding card
Hi,
Given that the Digium transcoding card has no external connections
(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
Does such a beast exist, or is it likely to? Am I correct in assuming
that this is a Digium-only product, and there is no OEM equivalent
"generic" board out there that I could be investigating? It would be
such a shame to waste a PCI slot that
2008 Nov 04
5
VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing
This Friday's edition of the weekly VoIP Users Conference call is all
about wideband audio (aka HD Voice) and conferencing. The guest for
this call is David Frankel, CEO of ZipDX a commercial service that
specializes in wideband conferencing. We expect an interesting call
touching on many aspects of VoIP going beyond the traditional phone
service, conference bridges, technical standards,
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
using asterisk 1.8.32.3
I am not able to make a call with video support. I do not know what I am
missing to make this video call.
Codec h264 should be supported.
sip*CLI> core show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME
2012 Sep 25
1
is silk included in asterisk 11?
I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see
silk listed in menuselect as a codec. But I also don't see an asterisk
11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk.
Do we use the asterisk 10 codec_silk.so ?
sean
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
Hello
in sip.conf I have ;
videosupport=yes
Kind regards.
J.
On 20-04-17 13:09, Marcelo Terres wrote:
> I suppose that you enable the video support on sip.conf, right?
>
> Regards,
> Marcelo H. Terres <mhterres at gmail.com>
> IM: mhterres at jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
>
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
2017 Apr 12
2
More issues with Siren14 datalen == 0 packets
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
uint32 = 2156475160, pad = "\030\063\211\200\000\000\000"}, delivery = {
tv_sec =
2014 Jan 23
1
mixmonitor extension
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---------------------------------------
Marek Cervenka
=======================================
2007 Dec 24
2
SIP Conference phones
Greetings list,
Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc..
A cursory googling has led me to the Polycom Soundpoint IP4000 at around the ?450 mark - any thoughts on this?
If anyone knows a good Polycom wholesaler in the UK, I'd be
2008 Sep 28
1
G.722 between Eyebeam and a Polycom IP650
Hi All,
So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes as "HD"
mode and the audio quality is certainly very good.
However, things are not so easy with Eyebeam and the
2011 Sep 30
1
Core show translation > 4000ms
Hi list,
we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is
Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk
1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both
machines for meetme timing.
Doing core show translation give on the Lenny server
Translation times between formats (in microseconds) for one
second of data
2017 Apr 12
2
More issues with Siren14 datalen == 0 packets
> All patches need to go into JIRA with a license agreement to be
> accepted.
Understood, but I was using it as an illustration. Note, however, that,
from a legal perspective, a patch such as this has no protectable IP (you
can't copyright the only way of doing something) and the GNU projects have
a formal rule that sufficiently-small patches need no assignments for that
reason, which