[asterisk-bugs] [JIRA] (ASTERISK-19871) No translation path between various SILK sample rates
Sean Bright (JIRA)
noreply at issues.asterisk.org
Mon Jan 20 14:26:25 CST 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-19871?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249488#comment-249488 ]
Sean Bright edited comment on ASTERISK-19871 at 1/20/20 2:25 PM:
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I am not able to reproduce this with Asterisk 13.30.0 and {{codec_silk}} 13.0_1.0.3:
{noformat}
*CLI> core show translation
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 silk8 silk12 silk16 silk24
slin8 - 8000 8000 8000 8000 8000 8000 8000 8000 6000 14000 14000 14000
slin12 8500 - 8000 8000 8000 8000 8000 8000 8000 14500 6000 14000 14000
slin16 8500 8500 - 8000 8000 8000 8000 8000 8000 14500 14500 6000 14000
slin24 8500 8500 8500 - 8000 8000 8000 8000 8000 14500 14500 14500 6000
slin32 8500 8500 8500 8500 - 8000 8000 8000 8000 14500 14500 14500 14500
slin44 8500 8500 8500 8500 8500 - 8000 8000 8000 14500 14500 14500 14500
slin48 8500 8500 8500 8500 8500 8500 - 8000 8000 14500 14500 14500 14500
slin96 8500 8500 8500 8500 8500 8500 8500 - 8000 14500 14500 14500 14500
slin192 8500 8500 8500 8500 8500 8500 8500 8500 - 14500 14500 14500 14500
silk8 9000 17000 17000 17000 17000 17000 17000 17000 17000 - 23000 23000 23000
silk12 17500 9000 17000 17000 17000 17000 17000 17000 17000 23500 - 23000 23000
silk16 17500 17500 9000 17000 17000 17000 17000 17000 17000 23500 23500 - 23000
silk24 17500 17500 17500 9000 17000 17000 17000 17000 17000 23500 23500 23500 -
{noformat}
Note that {{codec_resample}} must be loaded.
was (Author: seanbright):
I am not able to reproduce this with Asterisk 13.30.0 and {{codec_silk}} 13.0_1.0.3:
{noformat}
*CLI> core show translation
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 silk8 silk12 silk16
slin8 - 8000 8000 8000 8000 8000 8000 8000 8000 6000 14000 14000
slin12 8500 - 8000 8000 8000 8000 8000 8000 8000 14500 6000 14000
slin16 8500 8500 - 8000 8000 8000 8000 8000 8000 14500 14500 6000
slin24 8500 8500 8500 - 8000 8000 8000 8000 8000 14500 14500 14500
slin32 8500 8500 8500 8500 - 8000 8000 8000 8000 14500 14500 14500
slin44 8500 8500 8500 8500 8500 - 8000 8000 8000 14500 14500 14500
slin48 8500 8500 8500 8500 8500 8500 - 8000 8000 14500 14500 14500
slin96 8500 8500 8500 8500 8500 8500 8500 - 8000 14500 14500 14500
slin192 8500 8500 8500 8500 8500 8500 8500 8500 - 14500 14500 14500
silk8 9000 17000 17000 17000 17000 17000 17000 17000 17000 - 23000 23000
silk12 17500 9000 17000 17000 17000 17000 17000 17000 17000 23500 - 23000
silk16 17500 17500 9000 17000 17000 17000 17000 17000 17000 23500 23500 -
{noformat}
Note that {{codec_resample}} must be loaded.
> No translation path between various SILK sample rates
> -----------------------------------------------------
>
> Key: ASTERISK-19871
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-19871
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Codecs/codec_silk
> Affects Versions: 10.4.0, 11.5.1, 13.18.4
> Environment: Ubuntu - Kernel Version - 2.6.38-8-virtual , Asterisk Version - 10.4.0, Intel(R) Xeon(R) CPU E5506 @ 2.13GHz
> Reporter: Bala
> Attachments: codecs.txt, sip.txt
>
>
> Hi ,
> We are facing new issue that we create two users with enabled codec SILK24 and SILK8, when we calling to user1(SILK24) from user2(SILK8) we got the messages in Asterisk Console like this
>
> [May 14 16:17:35] WARNING[16123] chan_sip.c: Asked to transmit frame type silk8, while native formats is (silk24) read/write = silk8/silk8
> [May 14 16:22:11] WARNING[15914] channel.c: Unable to find a codec translation path from (silk24) to (silk8)
> This messages continuously running until we hanged up the call .
> But It is working fine with other codecs For example I can make the outgoing calls through my SIP provider (ULAW) and other codecs also
> Working Scenarios:
> SILK8 - ULAW,ilbc,gsm..
> SILK8 - SILK8
> Not Working
> SILK8 <-> SILK(16,24)
>
> Here is the table value that we got running Asterisk CLI command # core show translations.
> {noformat}
> gsm ulaw alaw g726 adpcm slin lpc10 ilbc silk8 silk12 silk16 silk24
> gsm - 15000 15000 15000 15000 9000 15000 15000 15000 15000 15000 15000
> ulaw 15000 - 9150 15000 15000 9000 15000 15000 15000 15000 15000 15000
> alaw 15000 9150 - 15000 15000 9000 15000 15000 15000 15000 15000 15000
> g726 15000 15000 15000 - 15000 9000 15000 15000 15000 15000 15000 15000
> adpcm 15000 15000 15000 15000 - 9000 15000 15000 15000 15000 15000 15000
> slin 6000 6000 6000 6000 6000 - 6000 6000 6000 6000 6000 6000
> lpc10 15000 15000 15000 15000 15000 9000 - 15000 15000 15000 15000 15000
> ilbc 15000 15000 15000 15000 15000 9000 15000 - 15000 15000 15000 15000
> silk8 15000 15000 15000 15000 15000 9000 15000 15000 - - - -
> silk12 15000 15000 15000 15000 15000 9000 15000 15000 - - - -
> silk16 15000 15000 15000 15000 15000 9000 15000 15000 - - - -
> silk24 15000 15000 15000 15000 15000 9000 15000 15000 - - - -
> {noformat}
> Please help to resolve this issue and also what I need to do resolve this issue?
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