asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	ee21eee7e0	Cache string values of formats on ast_format_cap() to save processing. Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c977f73e13	Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set. The callback function for changing the media address in streams wrongly assumes that a connection line will always be present. This is false as no line is present if a stream has been rejected. (closes issue ASTERISK-22645) Reported by: Rusty Newton ........ Merged revisions 400360 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	391f0003c4	Change the "external_media_address" PJSIP endpoint option to "media_address". The endpoint option does not apply to communication with external entities. Rather, the option is applied to all communications with the endpoint. The external_media_address transport configuration option may override the endpoint option if it turns out that we are going to be communicating with an external entity. Two things of note: 1) I have not updated the XML documentation. This is being taken care of by Rusty as part of his work on issue ASTERISK-22405 2) This commit is likely to cause testsuite failures since there are tests that use the external_media_address endpoint option, and they will need to be changed over. Well, I'm planning to get that updated ASAP after this commit. (closes issue ASTERISK-22528) reported by Rusty Newton ........ Merged revisions 399283 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4d348e853c	Add pass through support for Opus and VP8; Opus format attribute negotiation This patch adds pass through support for Opus and VP8. That includes: * Format attribute negotiation for Opus. Note that unlike some other codecs, the draft RFC specifies having spaces delimiting the attributes in addition to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in chan_sip, so a small tweak was also included in this patch for that. * A format attribute negotiation module for Opus, res_format_attr_opus * Fast picture update for VP8. Since VP8 uses a different RTCP packet number than FIR, this really is specific to VP8 at this time. Note that the format attribute negotiation in res_pjsip_sdp_rtp was written by mjordan. The rest of this patch was written completely by Lorenzo Miniero. Review: https://reviewboard.asterisk.org/r/2723/ (closes issue ASTERISK-21981) Reported by: Tzafrir Cohen patches: asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_" are now "chan_pjsip_" * All files that were "res_sip" are now "res_pjsip" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_" are now "pjsip_" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

5 Commits (2966ca288c2d0b97199c09f37e8d3f8cb6b94638)