[asterisk-users] AMR Codec
Has anyone successfully compiled the AMR codec into an Asterisk install, and if so, what steps did you take? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR Codec
On Wed, 2010-08-25 at 14:04 -0400, Matt wrote: Has anyone successfully compiled the AMR codec into an Asterisk install, and if so, what steps did you take? -- _ Just noticed that packman has prebuild packages: (with src-rpm's) amrnb Adaptive Multi-Rate Narrow-Band decoder and encoder library. (3GPP TS 26.104 V 7.0.0) amrwb Adaptive Multi-Rate Wideband decoder and encoder library. (3GPP TS 26.204 V7.0.0) opencore-amr Library of OpenCORE Framework implementation of Adaptive Multi Rate Narrowband and Wideband speech codec. codecs (and legal notice msg) come from: http://www.penguin.cz/~utx/amr hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR codec for Asterisk 1.6.1.X
Dear list, i have re-compiled again the source code of amr patch for 1.6 (https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/asterisk-1.6-AMR.patch) The patch does not compile with the static function into frame.c called : static int amr_samples(unsigned char *data, int datalen) i have removed the static and used like int amr_samples(unsigned char *data, int datalen) Anyone else got this issue ??? In this way the patch compile . It also show right format name when i try lo load codec_amr.so load codec_amr.so The 'load' command is deprecated and will be removed in a future release. Please use 'module load' instead. == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format amr to slin, cost 2000 == Registered translator 'lintoamr' from format slin to amr, cost 17997 Loaded codec_amr.so = (AMR Coder/Decoder) Also i have into the config file asterisk.conf the following value to filed transcode_via_sln = yes so transcode_via_sln = yes If i try to make a call to echotest by dialing 600 '600' = 1. Answer() [pbx_config] 2. Playback(demo-echotest) [pbx_config] 3. Echo() [pbx_config] 4. Playback(demo-echodone) [pbx_config] with a client that have only enabled amr codec i got this output: [May 6 17:51:11] WARNING[9684]: chan_sip.c:7654 process_sdp: Unsupported SDP media type in offer: audio 4002 RTP/SAVP 114 18 113 0 8 101 Anyone know how to get this AMR codec doing transcoding on asterisk 1.6? Many thanks in advantage Andrea Il 05/05/2010 18:13, Adrian Marsh ha scritto: It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*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 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats
[asterisk-users] AMR codec for Asterisk 1.6.1.X
Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*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 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR codec for Asterisk 1.6.1.X
== Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Probably shouldn't be listing it as unknown Have you tried using that AMR codec beyond commands in the asterisk cli? Did the patch apply cleanly? On Wed, May 5, 2010 at 6:21 AM, Andrea Cristofanini andrea.cristofan...@zerozero39.it wrote: Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*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 slin16 g723 - - - - - - - - - - - - - - gsm - - 2 2 2 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 1 2 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 - 2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2 - 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 2 2 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 1 1 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - - - - - - - - - - - - ilbc - - - - - - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 12001 12001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audio gsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8) (0x100) audio g729 (G.729A) 512 (1 9) (0x200) audio speex (SpeeX) 1024 (1 10) (0x400) audio ilbc (iLBC) 2048 (1 11) (0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] AMR codec for Asterisk 1.6.1.X
It says in the readme from that link you provided: This patch adds AMR-NB support to Asterisk 1.4 (for Asterisk 1.6 check out asterisk 1.6 branch and use the asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov)) Did you use the 1.6 branch and patch ?? I'll have to try this myself at some point. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrea Cristofanini Sent: 05 May 2010 14:22 To: asterisk-users@lists.digium.com Subject: [asterisk-users] AMR codec for Asterisk 1.6.1.X Hi list, Anyone have successfully compiled amr codec for asterisk 1.6.1.X ? I still have no problem compiling and playing with it on Asterisk 1.4.X. I have used the following patch : https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/ Hare is what i get while loading codec_amr.so debbi*CLI load codec_amr.so == Parsing '/etc/asterisk/codecs.conf': == Found -- codec_amr: parsing codecs.conf -- codec_amr: set octed-aligned mode to 1 -- codec_amr: set dtx mode to 0 -- codec_amr: AMR mode set to MR122 (7) codec_amr: enc_mode = 7, dtx = 0 == Registered translator 'amrtolin' from format unknown to slin, cost 4000 == Registered translator 'lintoamr' from format slin to unknown, cost 32002 Loaded codec_amr.so = (AMR Coder/Decoder) debbi*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 slin16 g723 - - - -- - - - - - - - - - gsm - - 2 22 2 1 4001 12002 - - 2 2 4003 ulaw - 12002 - 12 2 1 4001 12002 - - 2 2 4003 alaw - 12002 1 -2 2 1 4001 12002 - - 2 2 4003 g726aal2 - 12002 2 2- 2 1 4001 12002 - - 2 2 4003 adpcm - 12002 2 22 - 1 4001 12002 - - 2 2 4003 slin - 12001 1 11 1 - 4000 12001 - - 1 1 4002 lpc10 - 16001 4001 4001 4001 4001 4000 - 16001 - - 4001 4001 8002 g729 - 16001 4001 4001 4001 4001 4000 8000 - - - 4001 4001 8002 speex - - - -- - - - - - - - - - ilbc - - - -- - - - - - - - - - g726 - 16001 4001 4001 4001 4001 4000 8000 16001 - - - 4001 8002 g722 - 20001 8001 8001 8001 8001 8000 12000 20001 - - 8001 - 4001 slin16 - 24001 12001 1200112001 12001 12000 16000 24001 - - 12001 4000 - debbi*CLI core show file formats version debbi*CLI core show co codec codecs config debbi*CLI core show code codecs codec debbi*CLI core show codec codecs codec debbi*CLI core show codec audio Usage: core show codec number Displays codec mapping debbi*CLI core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) debbi*CLI The CLI does not show codec audio or codedc translation for AMR NB. Anyone have any idea ?? Thanks in advantage Andrea -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On 17 Sep 2006, at 23:04, Net Nut wrote: Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Probably not. Unless you live in a country where the patent isn't valid. You'd have to read the license terms from the patent holder(s) to be sure. On the other hand I doubt that they would go after you for a single channel. Conversely, you should consult the licensing folks at Digium to see if adding a module that contains a patented codec is ok with them. You certainly can't distribute the result in any way without breaching the GPL/Digium's license. Likewise no one can help you building it since you may not distribute the sourcecode either. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On Sun, Sep 17, 2006 at 03:04:35PM -0700, Net Nut wrote: Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Voiceage are quite agressive in terms of licensing. However as an individual it's probably not worth their efforts to do anything as the results wouldn't be worth it. If you run a business and the business has assets, then it's a different matter. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Tim Panton wrote: On 16 Sep 2006, at 20:38, Net Nut wrote: So with that said, can anyone recommend a way that I can get a sip client on a cell phone that uses H.263 and amr to talk to an asterisk system? Is it just not possible because of licensing? It sounds kind of lame to have a sip client that can't talk to anything else because of codecs.. Well Asterisk does not _have_ to have an amr codec for you to be able to use your handset. If you have several of these handsets or other devices that support amr, then asterisk can route calls between them, just passing the stream through. If you want any of the interesting asterisk features, then it will need to transcode, and then Steve's right, not only do you have to add codec code to Asterisk (which is almost certainly a GPL violation) you also have to pay the patent holder for any commercial use of the codec. Your best hope is if a few of us can persuade Digium to support amr in the same way that they license g729 . Tim. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
So with that said, can anyone recommend a way that I can get a sip client on a cell phone that uses H.263 and amr to talk to an asterisk system? Is it just not possible because of licensing? It sounds kind of lame to have a sip client that can't talk to anything else because of codecs.. Steve Kennedy wrote: On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
On 16 Sep 2006, at 20:38, Net Nut wrote: So with that said, can anyone recommend a way that I can get a sip client on a cell phone that uses H.263 and amr to talk to an asterisk system? Is it just not possible because of licensing? It sounds kind of lame to have a sip client that can't talk to anything else because of codecs.. Well Asterisk does not _have_ to have an amr codec for you to be able to use your handset. If you have several of these handsets or other devices that support amr, then asterisk can route calls between them, just passing the stream through. If you want any of the interesting asterisk features, then it will need to transcode, and then Steve's right, not only do you have to add codec code to Asterisk (which is almost certainly a GPL violation) you also have to pay the patent holder for any commercial use of the codec. Your best hope is if a few of us can persuade Digium to support amr in the same way that they license g729 . Tim. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amr codec
Steve Kennedy wrote: On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. While the Voiceage licence for G.729 seems to cover you for patent issues on that codec, their licence for AMR does not. Their patent pool does not include all those who claim a patent on AMR. So, you need their licence plus you need to search out and licence other people's patents too. I assume this is why the AMR licence appears cheaper than the G.729 one. Steve' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] amr codec
I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. Thanks, Grant ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
