| Summary: | ASTERISK-15053: [patch] Extend slin16 support to SIP calls | ||
| Reporter: | Ken Fister (kfister) | Labels: | |
| Date Opened: | 2009-10-29 06:59:16 | Date Closed: | 2010-06-17 13:36:07 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/NewFeature |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 1.6.2.0-rc3-slin16.patch ( 1) 16153-1.6.2.0-rc5.patch ( 2) slin16.sip.patch.1 | |
| Description: | 1.6.2.0-rc3 appears to support using the slin16 codec for IAX2 calls. These changes extend the functionality to SIP calls. chan_sip must be told to "allow=slin16." I have tested this on my own system. It works with Aastra 57i telephone running firmware 2.5.2.1010. As of this week SVN 1.6.2 and SVN trunk are also missing this functionality, similar changes in the relevant files (rtp_engine.c instead of rtp.c) should implement it. ****** ADDITIONAL INFORMATION ****** main/rtp.c: support slin16 struct ast_frame: call ast_frame_byteswap_be for AST_FORMAT_SLINEAR16, as is done with AST_FORMAT_SLINEAR struct mimeType: associate AST_FORMAT_SLINEAR16 with L16 @ 16000 Hz Other changes are "cosmetic" main/frame.c: reveal new codecs in "core show codecs" show_codecs(): change i<13 to i<16 in for loop so that siren7, siren14, and slin16 codecs are listed. channels/chan_sip.c: process_sdp(): Add the sample rate to the "Found audio description format" debug messages. This is useful because it shows the sample rate that has been read from the SDP and given to ast_rtp_set_rtpmap_type_rate(). | ||
| Comments: | By: Ken Fister (kfister) 2009-11-12 19:36:51.000-0600 Cleaned up patch (sorry I'm new at making patch files) My own testing has proceeded with 1.6.2.0-rc4 and -rc5 with satisfactory results By: Ken Fister (kfister) 2009-11-25 00:30:44.000-0600 Still okay with 1.6.2.0-rc6 By: Tilghman Lesher (tilghman) 2010-05-27 10:50:48 Patches for new features must be for trunk, not 1.6.2. In addition, slin16 is only meant to be used internally as an intermediate step for transcoding 16kHz audio. Additionally, you've specified the same text description for slin16 as for slin, which means the codec will never be sourced. By: Paul Belanger (pabelanger) 2010-05-28 11:30:49 This patch also fails the CODING-GUIDELINES. IE: Spacing between variables, and braces. By: Paul Belanger (pabelanger) 2010-06-10 15:05:22 Feel free to reopen when you are ready to submit your patch. --- Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines By: Malcolm Davenport (mdavenport) 2010-06-15 09:08:55 Patch provided for trunk. Tested using Xlite 3.0. By: Digium Subversion (svnbot) 2010-06-17 13:36:06 Repository: asterisk Revision: 271261 U trunk/CHANGES U trunk/main/rtp_engine.c U trunk/res/res_rtp_asterisk.c ------------------------------------------------------------------------ r271261 | dvossel | 2010-06-17 13:36:06 -0500 (Thu, 17 Jun 2010) | 9 lines adds support for slin16 in sip (closes issue ASTERISK-15053) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=271261 | ||