Posts Tagged codecs

Fax For Asterisk 1.1.6 Release Announcement

Posted by admin in asterisk, codecs, digium, g.711, t.38 on January 7, 2010

T.38 fax for Asterisk

T.38 fax for Asterisk

Digium is pleased to announce the release of version 1.1.6 of its Fax For Asterisk product, a commercial grade FAX add-on module for open source Asterisk.

This release contains a number of significant improvements, including:

  • Support for 64-bit Linux installations.
  • Reduction in resource consumption, and increase in performance, of T.38 session handling.
  • Simplification of session handling during transition from G.711 to T.38 mode.
  • Adoption of the latest Asterisk T.38 negotiation API, ensuring interoperability with a wide range of T.38 endpoints.

Version 1.1.6 of Fax For Asterisk is available for immediate download at http://www.digium.com/en/docs/FAX/faa-download.php. Note that because this release uses the very latest T.38 negotiation mechanism in Asterisk, it is not compatible with all released versions of Asterisk. The Fax For Asterisk download selector lists the Asterisk versions supported by this release.

For more information about Fax For Asterisk, please visit the product page.

Thank you for your support!
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Digium Releases TCE400B PCI-Express Transcoder Card

Posted by admin in codecs, digium, g.711, g.723.1, g.729, hardware on June 30, 2009

Digium TCE400B

Digium TCE400B

Digium releases the TCE400B PCI Express card for use with voice applications based on the open source Asterisk telephony platform. The new card provides hardware-based voice compression and decompression (codec) capabilities to shift transcoding from software to hardware. Using the TCE400B in place of a software-only solution places fewer demands on servers and frees up Asterisk to more efficiently process calls and to provide functionality for phone systems such as call recording, conference calling and interactive voice response.

The TCE400B is a half-length, low-profile PCI-Express x1 card for transforming complex VoIP codecs into simple codecs. This product is, essentially, a PCI-Express version of Digium’s existing TC400B product.

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G.729 and G.723.1 codecs x86 and x86_64 Linux and FreeBSD binaries for Asterisk open source PBX

Posted by admin in codecs, g.723.1, g.729 on June 24, 2009

DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm.

Sources

To compile the codecs you need Intel IPP libraries installed. Currently only Asterisk 1.4, 1.6 and TRUNK are supported. Support for Asterisk 1.2 and Callweaver is planned, for now use the binaries. Use “g723 debug” and “g729 debug” commands to print statistics about received frame sizes, can aid in debugging audio problems. You need to bump Asterisk verbosity level to 3 to see the numbers.

Binaries

  • choose codec binary appropriate for your Asterisk version and CPU type, use x86_64 for 64-bit mode, scroll to the end of the list for FreeBSD binaries
  • delete old codec_g729/723*.so files (if any) from /usr/lib/asterisk/modules directory
  • copy new codec_g729/723*.so files into /usr/lib/asterisk/modules directory
  • restart Asterisk
  • check the codec is loaded with ‘core show translation recalc 10′ on Asterisk console (‘show translation’ in Asterisk 1.2)
  • G.723.1 send rate is configured in Asterisk codecs.conf file (Linux Asterisk 1.2, 1.4, 1.6, TRUNK and Callweaver, FreeBSD 7.x Asterisk 1.4 binaries only):
    [g723]
; 6.3Kbps stream, default
sendrate=63
; 5.3Kbps
;sendrate=53

    This option is for outgoing voice stream only. It does not affect incoming stream that should be decoded automatically whatever the bitrate is.

  • in sip.conf or/and iax.conf configure the codec(s) either globally or under respective peer, for example:
    disallow=all
allow=g729
  • for detailed information about Asterisk configuration visit voip-info.org
  • in case of problems read Notes and Troubleshooting

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