Summary:	ASTERISK-26564: codec_silk: Very poor sound quality in SiLK implementation
Reporter:	Samuel For (samfun)	Labels:
Date Opened:	2016-11-07 17:36:02.000-0600	Date Closed:	2016-12-02 10:13:28.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Codecs/codec_silk
Versions:	13.12.1	Frequency of Occurrence	Constant
Related Issues:
Environment:	Ubuntu 14.04. Asterisk 13.12.1.	Attachments:	( 0) 02_silkTraud_ast13.7.2.pcap ( 1) 02_silkTraud_ast13.7.2.wav ( 2) 04_silkDigium_ast13.12.1.pcap ( 3) 04_silkDigium_ast13.12.1.wav
Description:	Using Asterisk 13.7.2 with the traud/asterisk-silk patch gives very good sound quality on transcoded audio. Using Asterisk 13.12.1 with the digium-silk-binary gives very poor sound  quality on transcoded audio. The quality difference can easily be heard by listening to the two different sound files I've attached. I have attached the following to this ticket: 1) PCAPs for the calls on versions 13.7 and 13.12.1 2) Audio file recordings for each version I used SLN16 sounds version 1.4.27 for both tests. I used the following dialplan: exten =>       5004,n,Answer exten =>       5004,n,Wait(3) exten =>       5004,n,Playback(priv-callee-options) exten =>       5004,n,Hangup I used Jitsi as a client with SiLK/16000. This was the codec setting for codecs.conf for 13.12.1: [silk16] type=silk samprate=16000 fec=true packetloss_percentage=10 maxbitrate=24000 dtx=true As it stands now, the Digium SiLK codec binary is unusable unfortunately.
Comments:	By: Asterisk Team (asteriskteam) 2016-11-07 17:36:03.342-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Samuel For (samfun) 2016-11-07 17:38:22.936-0600 PCAPs and sound recordings showing the quality difference. The Traud version can be found at https://github.com/traud/asterisk-silk. By: George Joseph (gjoseph) 2016-11-15 16:56:39.088-0600 Hi Sam, Can you do a test call again and while the call is in progress, do a "core show channel <channel name>"?  You can do a "core show channels" to get the specific channel or hit <TAB> after typing "core show channel ".  I want to see what path the transcoding is taking. Does the issue only happen when transcoding to/from specific formats? How about phone-to-phone or local only? Finally is this chan_sip or chan_pjsip? thanks! By: George Joseph (gjoseph) 2016-11-15 17:24:50.253-0600 Oh and as a side note...   The Digium codec_silk doesn't read codecs.conf at all. By: Samuel For (samfun) 2016-11-18 07:54:46.472-0600 This was chan_sip. I will try to get the output of those commands next week. We have already torn down this environment so I will need to rebuild it and run the tests again. If the codecs.conf is not read for silk then this file needs an update: https://github.com/asterisk/asterisk/blob/master/configs/samples/codecs.conf.sample By: Samuel For (samfun) 2016-12-02 07:56:37.586-0600 Hi again, We have decided internally to not use SiLK in the project. The prototype environment which we used for this has already been torn down. I would suggest that this ticket can be closed unless someone else wants to progress it. By: Joshua C. Colp (jcolp) 2016-12-02 10:13:28.601-0600 Suspended per reporter since further details from the environment can not be retrieved and since SILK will not be used. By: Richard Mudgett (rmudgett) 2016-12-02 10:38:26.526-0600 From the attached wav file I suspect that the default GSM sound files were installed and got translated to SILK. By: Samuel For (samfun) 2016-12-02 13:36:07.996-0600 @Richard: I don't think it explains why the quality differs. The installation script was exactly the same except for the patch being applied / not being applied. By: Asterisk Team (asteriskteam) 2016-12-02 13:36:08.185-0600 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.