Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

On Sun, Dec 06, 2015 at 08:42:42PM -0600, Matthew Jordan wrote: > > pass-through including fmtp negotiation (level 1) > > | pass-through plus library detection in ./configure (level 2) > > | | transcoding module in codecs/codec_* (level 3) > > | | | > > [x] [x] [x] Codec 2

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

> If I get a "codec2" stream, which rate (and/or other parameters) are used? Faced the same question, when I started with Codec 2. I am glad, somebody is interested. I am going to add this to the Read Me in my GitHub repository: Currently, FreeSWITCH and CSipSimple support only the first Codec 2

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

On Tue, Dec 8, 2015 at 4:17 AM, Alexander Traud wrote: > >> Question #1: Level of Integration? > >> [x] [x] [ ] GSM-EFR; GSM-FR is available already > > More analysis would have to be done for a codec module. > > GSM-EFR uses the same library as AMR. Therefore, it is

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

On Tue, Nov 24, 2015 at 9:08 AM, Alexander Traud wrote: > Thanks to the codec/format changes which were introduced with Asterisk 13, > adding new trancoding modules is possible within one working day. Thanks to > format-attribute modules, the debugging of the

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

> If you need testing, I can support you. Thanks for the offer! Just give them a try and report all issues via their GitHub repository or privately, as you like. By the way just to avoid a misunderstanding: All five modules are finished and passed several tests of my own. They all support

Re: [asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

Hello Alexander, great work!! If you ask, it would be nice if transcoding support is there for GSM-EFR and both AMR Codes. If you need testing, I can support you. Kind regards, André Am 24.11.2015 um 16:08 schrieb Alexander Traud: > Thanks to the codec/format changes which were introduced

[asterisk-dev] Transcoding: Codec 2, iLBC 20, SILK, GSM-EFR, AMR(-WB)

Thanks to the codec/format changes which were introduced with Asterisk 13, adding new trancoding modules is possible within one working day. Thanks to format-attribute modules, the debugging of the SDP/fmtp-negotiation resides in one source file. Therefore, I was able to port five formats to