Displaying 19 results from an estimated 19 matches for "speex16".
2005 Oct 04
1
Strange Problem
...n I
tried to integrate the wideband mode. But the program
crashes mysteriously. My encode and decode codes for
wide band mode are exact similiar to that of
narrowband, except the mode initialization, where I
put "speex_wb_mode" instead of "speex_nb_mode".
My encoding code:
bool
SPEEX16::Encode(const std::string& raw, std::string&
encoded)
{
speex_bits_reset(&bits);
speex_encode_int(state, (short*)raw.data(), &bits);...
2012 Mar 21
0
AMR Codec with Asterisk 1.8.9.1
...risk.
*> 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 amr siren14 slin16 g719 speex16 siren7 testlaw
g723 - - - - - - - - -
- - - - - - - - - - -
gsm - - 1001 1001 - - 1000 - -
10999 - - - *9998* - - - - - 2000...
2014 Feb 11
0
g726 transcoding
...alaw To adpcm : No Translation Path
alaw To slin : (alaw)->(slin)
alaw To lpc10 : No Translation Path
alaw To g729 : No Translation Path
alaw To speex : (alaw)->(slin)->(speex)
alaw To speex16 : (alaw)->(slin)->(slin16)->(speex16)
alaw To ilbc : No Translation Path
alaw To g726aal2 : No Translation Path
alaw To g722 : No Translation Path
alaw To slin16 : (alaw)->(slin)->(slin16)
alaw...
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
...type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm
[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes
[2001]
type=auth
auth_type=userpass
password=test
username=test
Best Regards,
Madush...
2019 Jul 05
4
Asterisk and Linphone
Hi all - I am using asterisk 13.27.0 with Linphone.
I turned off all codes on linphone except the one I want to try. For
example:
opus and speex (so only one enabled at a time).
Then did this same on asterisk for the linphone extension.
disallow=all
allow=speex
(for example).
Then I place my call and the call fails. if I enable something like gsm,
ulaw, alaw the call works fine. Why does the
2012 Nov 21
1
core show translation - difference in Asterisk Versions
...60 or 1 seconds but nothing changes
on Asterisk 11 (VM, Cloud or even physical machine). Is it slin?, adding
this overhead or there is something I am overlooking?.
*
*
*Asterisk 11.0.1 => core show translation **(in microseconds)*
*gsm ulaw alaw g726 adpcm slin lpc10 g729 speex speex16
ilbc g726aal2 g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44
slin48 slin96 slin192*
*gsm *- 15000 *15000 *15000 15000 9000 15000 15000 *15000 *23000
15000 15000 17250 17000 15000 23000 17000 17000 17000 17000
17000 17000 17000
*ulaw *15000 - 9150 15...
2016 Sep 09
2
Asterisk 13 PJSIP with Snom 710
...gt; allow=alaw
> transport=system-udp
> auth=2001
> aors=2001
> direct_media=no
> rtp_symmetric=yes
> force_rport=yes
> allow=alaw
> allow=speex
> allow=speex16
> allow=speex32
> allow=gsm
>
>
> [2001]
> type=aor
> qualify_frequency=5000
> authenticate_qualify=yes
> max_contacts=1
> remove_existing=yes
>
> [2001]
>...
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
...;< 30) (0x40000000) (unk) unknown
(unknown)
2147483648 (1 << 31) (0x80000000) (unk) unknown
(unknown)
4294967296 (1 << 32) (0x100000000) audio g719
(ITU G.719)
8589934592 (1 << 33) (0x200000000) audio speex16
(SpeeX 16khz)
17179869184 (1 << 34) (0x400000000) audio unknown
(unknown)
34359738368 (1 << 35) (0x800000000) audio unknown
(unknown)
68719476736 (1 << 36) (0x1000000000) audio unknown
(unknown)
137438953472 (1...
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
...(unknown)
>> 2147483648 (1 << 31) (0x80000000) (unk) unknown
>> (unknown)
>> 4294967296 (1 << 32) (0x100000000) audio g719 (ITU
>> G.719)
>> 8589934592 (1 << 33) (0x200000000) audio speex16
>> (SpeeX 16khz)
>> 17179869184 (1 << 34) (0x400000000) audio unknown
>> (unknown)
>> 34359738368 (1 << 35) (0x800000000) audio unknown
>> (unknown)
>> 68719476736 (1 << 36) (0x1000000000)...
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
...1 asterisk asterisk 282 2011-10-11 08:06 msg0008.txt
-rw-rw---- 1 asterisk asterisk 55724 2011-10-11 08:06 msg0008.wav
-rw-rw---- 1 asterisk asterisk 5715 2011-10-11 08:06 msg0008.WAV
Codec negotiation:
Capabilities: us - 0x80030c7fffff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719),
peer - audio=0xc (ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0
(nothing), combined - 0x38000c (ulaw|alaw|h263|h263p|h264)
In asterisk.conf we even activate
transcode_via_sln = yes ;Build...
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
...Addr->IP : 192.168.0.102:5060*
> Defaddr->IP : (null)
> Prim.Transp. : UDP
> Allowed.Trsp : UDP
> Def. Username: 6201
> SIP Options : (none)
> Codecs : 0x80030c7fffff
> (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719)
> Codec Order : (none)
> Auto-Framing : No
> Status : Unmonitored
> Useragent :
> Reg. Contact :
> Qualify Freq : 60000 ms
>...
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
...0) (unk) unknown
> (unknown)
> 2147483648 (1 << 31) (0x80000000) (unk) unknown
> (unknown)
> 4294967296 (1 << 32) (0x100000000) audio g719
> (ITU G.719)
> 8589934592 (1 << 33) (0x200000000) audio speex16
> (SpeeX 16khz)
> 17179869184 (1 << 34) (0x400000000) audio unknown
> (unknown)
> 34359738368 (1 << 35) (0x800000000) audio unknown
> (unknown)
> 68719476736 (1 << 36) (0x1000000000) audio unknown
> (unkn...
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2014 Dec 11
0
PJSIP configuration question
...rmat 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729)
Non-codec capabi...
2013 Sep 26
0
Asterisk / SIP-Call / AGI-Script / SIGHUP after Answer
...o description format speex for ID 98
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 96
Capabilities: us - (gsm|ulaw|alaw|g729|g722), peer -
audio=(ulaw|alaw|speex16|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:10054
Looking for 3 in thorsten_sip_for_testing (domain myho...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2016 Dec 10
6
failing to start asterisk on centos7
...egistered 'audio' codec 'speex' at sample rate '8000' with id '19'
== Created cached format with name 'speex'
== Registered 'audio' codec 'speex' at sample rate '16000' with id '20'
== Created cached format with name 'speex16'
== Registered 'audio' codec 'speex' at sample rate '32000' with id '21'
== Created cached format with name 'speex32'
== Registered 'audio' codec 'ilbc' at sample rate '8000' with id '22'
== Created cached format wi...
2017 Apr 21
2
Asterisk 1.8.32.3 : no video (h.264)
...lt; 31) (0x80000000) (unk) unknown
>>>> (unknown)
>>>> 4294967296 (1 << 32) (0x100000000) audio g719
>>>> (ITU
>>>> G.719)
>>>> 8589934592 (1 << 33) (0x200000000) audio speex16
>>>> (SpeeX 16khz)
>>>> 17179869184 (1 << 34) (0x400000000) audio unknown
>>>> (unknown)
>>>> 34359738368 (1 << 35) (0x800000000) audio unknown
>>>> (unknown)
>>>>...
2011 Feb 10
2
Unable to make outgoing calls with Internode
...for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719),
peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100d0d
(g723|ulaw|alaw|g726|g729|ilbc|h263p)
Non-codec capabilities (dtmf): us - 0...