stream: correct Doxygen for peer field #2202
Merged
Conversation
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
alfredh
added a commit
that referenced
this pull request
Nov 1, 2022
* cmake: add opensles module (#2108) * test/call: Add test_call_change_videodir (#2080) * cmake: bump min version to 3.10 (#2112) * zrtp: remove module, use gzrtp instead (#2109) * Avoid gzrtp compile warnings (#2110) * Avoid gzrtp mdule unknown pragmas warnings * Disable more warnings * Avoid two unused param warnings * Update video in menu when UA_EVENT_CALL_REMOTE_SDP is recieved (#2113) * httpreq: update call to http_reqconn_set_body() with mbuf as parameter (#2100) Co-authored-by: Christian Spielberger <c.spielberger@commend.com> * call: send reinvite after established handlers (#2117) * cmake/modules: remove find_package ZRTP (unused) * ua,account,menu,test: refer out of dialog (#2115) * ua: add ua_handle_refer * ua: add UA_EVENT_REFER * ua: add Refer-Sub to REFER response * ua,menu: add API function and command for sending out-of-dialog REFER * ua: add mandatory headers to REFER Adds mandatory headers Supported and Contact. * ua,menu: do URI complete for refer-to URI * account: add UAS authentication Add account configuration for incoming SIP request authorization. * ua: add authorization check for incoming out-of-dialog REFER * ua: user@host for digest realm * ua: handle REFER set Refer-Sub defaul to true * ua: SIP request handler should not print warning and return false * test: add test for out-of-dialog REFER * remove unused functions in baresip.h (#2122) * webrtc: make https optional (#2120) Testing with localhost needs no SSL connection * Restored original working behavior in uag request_handler (#2124) Thanks for review. I'll do the merge then. * uag: out-of-dialog REFER handler checks to.tag (#2125) The REFER handler of uag has to check if the to.tag is unset. This means that the SIP request is outside of a dialog. Otherwise false should be returned and the processing is passed to the SIP session request handler in re listen.c which handles call transfers. * Update media fixes (#2116) * Revert "Update video in menu when UA_EVENT_CALL_REMOTE_SDP is recieved (#2113)" This reverts commit 344e053. * call: do not call update_media in progress handler It is called in offer/answer handlers. These are called now for each SDP offer/answer. * call: in update_media() stop video if not ready * menu: support also media dir changes during early state * account: set 100rel default to no (#2128) 100rel will not be put to the Supported header. And if the peer has 100rel in the Required header, baresip answers with a SIP 420 Bad Extension. * avcodec: remove usage of old FFmpeg api (before 4.1.9) (#2126) * rtp: Improve media synchronization (#2129) * avformat: remove usage of old FFmpeg api (#2130) * i2s: remove deprecated module (#2131) * ci: migrate to CMake (#2132) * menu: during early media switch on/off ringback (#2133) If remote informs about audio direction changes during early media state, the ringback might has to be turned on/off. If there is incoming early audio, ringback is turned off otherwise turned on. * cmake/FindREM: add rem-static target name * call, event, audio: send DTMF via hidden call (#2134) * call, event: add a UA event stop processing flag UA event handlers are processed in configuration order from h_1 to h_n. Any UA event handler h_x may set the calls event stop flag true. Following UA event handlers are not processed. The next UA event is processed again by the UA event handlers h_1 to h_x and not by h_{x+1} to h_n. * ua,reg,serreg: fix serial registration mode (#2139) The commit a2ab6ab changed the behavior of the serial mode. This adds a function `ua_stop_register()` that has the previous behavior. * opus/decode: fix possible overflow multiplication Multiplication result may overflow 'unsigned int' before it is converted to 'size_t'. * video: fix potentially uninitialized dir variable Detected by CodeQL. * webrtc/sess: fix potentially uninitialized type variable Detected by CodeQL * config: use secure fs_fopen fixes CodeQL warning: A file may be created here with mode 0666, which would make it world-writable. * cmake: set atomic-implicit-seq-cst only for C language * cmake: define -Wshorten-64-to-32 C only * Cmake of webrtc_aec module plus remove of unused aec.cpp var (#2144) * cmake: make include dir public * cmake: add APP_MODULES and APP_MODULES_DIR (#2148) * readme: add cmake examples * Added cmake of gst module (#2149) * Improved call closed message (#2151) * gtk & menu: Fix potential memory leaks (#2153) Fixes a few potential memory leaks in functions calling account_uri_complete. * call: allocate streams after peer_uri was set For now the peer_uri should be available when the streams are allocated. In a next PR it could be passed to the streams. Maybe in another PR the streams could be allocated after the SDP negotiation. Currently this is not possible due to the "common audio or video codecs" check which depends on the audio/video streams. * ua,call: allocate streams for handle OPTIONS The response to an out-of-dialog OPTIONS needs an SDP body which is generated by a dummy call and audio/video stream objects. * ua,call: got_offer cleanup * dshow/cmake: fix stdc++ linking with MSVC * cmake: fix MSVC library output name ref #2155 * cmake: build static by default on win32 * webrtc: add HAVE_GETOPT * config: ignore dirent.h on win32 win32 should be build static, so module counting is not needed. * ua: do not duplicate request URI parameters If ua_connect or ua_connect_dir are called and the request URI contains URI parameters, it could occur that some parameters are be duplicated if the account luri contained the exact same parameter. Now, only non-duplicate URI parameters are appended to the request URI in ua_connect_dir. * cmake: add netroam module (#2170) * cmake: add portaudio module (#2173) * cmake: add jack module (#2172) * avcodec,config: add setting for keyframe interval (#2171) * cmake: add sdl module (#2174) * call: set peer URI early for incoming calls (#2168) * cmake: Add options -DDEFAULT_CAFILE="…" and -DDEFAULT_AUDIO_DEVICE="…" (#2179) * cmake: add gtk module (#2176) * cmake: add opus_multistream module (#2175) * cmake: synchronize behaviour of -DSHARE_PATH="…" with GNU Makefiles (#2180) * cmake: synchronize behaviour of -DMOD_PATH="…" with GNU Makefiles (#2181) * Move docs/COPYING to LICENSE and update content to match with re/rem (#2188) * cmake: add ABI (soname) versioning (#2187) * misc: Use example domains and IPs (#2186) - RFC 2606 (Reserved Top Level DNS Names) - RFC 3849 (IPv6 Address Prefix Reserved for Documentation) - RFC 5737 (IPv4 Address Blocks Reserved for Documentation) * cmake: symlink modules (#2190) * cmake: add mpa module (#2191) * release v2.8.0 (#2193) * baresip.h: bump BARESIP_VERSION to v2.8.0 (#2196) * release v2.8.1 (#2197) * sndfile Module - filename includes strm->cname (i.e. call->local_uri)… (#2165) * log: optional timestamps (#2169) * avcodec: remove H263 codec (#2182) * mk: bump PROJECT_NUMBER in Doxyfile (#2201) * stream: correct Doxygen for peer field (#2202) * cmake: add pre version handling (#2203) * cmake,mk: bump dev version * bump dev version * cmake,debian: use dh-cmake (#2204) * cmake: add pkgconfig (#2205) * Avoid webrtc_aecm module C++20 extension warnings (#2215) * cmake/ctrld_dbus: ninja and subdirectory fixes (#2221) Fixes CMAKE_CURRENT_SOURCE_DIR and target name needs different to output name. * portaudio: hide alsa/jack and other driver debug messages on init * cmake: link CMAKE_CURRENT_BINARY_DIR modules (#2223) * cmake,debian: fix libbaresip dependency (#2224) * cmake: set C only flags (#2226) * FindPNG needs to find also include directory (#2230) * FindVPX needs to find also include directory (#2231) * Multicast send events on mcreg enable commands (#2219) * mc: send receiver stopped event on mcregen disable * mc: send event if receiver gets muted by prio enable/disable command * call, menu: support display name for outgoing calls (#2220) * call, menu: support display name for outgoing calls This enables parsing the display name out of the dial URI and report it via UA events to the application. * menu: support also short form for dialdir with display name * menu: ccheck fix * menu: dialdir fix long form * menu: support REFER with display name * call: hangup call on transp reset if necessary (#2229) * call: hangup call on transp reset if necessary If the transport is reset (e.g. an IP address change) and a call is in the early media state, session modifications might not be possible. In this case, the peer will never receive our transport update (e.g our new IP) and we will no longer be able to communicate with the peer. In such cases - when target refreshes are not possible - hangup the call. * uag & call: Move hangup on transp reset to uag * portaudio: add mediadev_add with mediadev driver fields (#2233) * baresip: extend mediadev struct with info fields * portaudio: add mediadev_add with mediadev info fields * improve generic and driver specific options * git: ignore generated cmake files * call: fix mnat call_streams_alloc (#2242) * jack: fix CodeQL uninitialized local variable (#2244) The variable `sampc_rs_out` may not be initialized here. * Avoid snapshot compiler warnings (#2239) * Avoid snapshot compiler warnings * Add Wno-clobbered if compiler is not Clang * avformat: remove old call to avcodec_register_all() (#2246) * auresamp: fix typo * update git ignore * avformat: remove LIBAVUTIL_VERSION_MAJOR check (#2247) * ua: wording for warning in ua_refer_send() (#2249) * ua: use mbuf functions for ua_connect_dir (#2250) * mk: update doxygen file * ci: use actions/checkout@v3 (#2254) * avcodec: remove av_packet_free() wrapper (#2255) * selfview: create window in encode_update (#2253) * selfview: create window in encode_update this fixes issues in sdl with starting timer from wrong thread * fix lint * Update selfview.c * config: default hwaccel for windows * avcodec: clean up old LIBAVCODEC_VERSION_INT * alsa: use C11 threads (#2256) * config: fix template for avcodec_xxx (#2258) * config: print default hwaccel * avformat: use C11 threads (#2259) * v4l2: use C11 threads (#2261) * avcodec: remove LIBAVUTIL_VERSION_MAJOR check (#2260) * avcodec: print supported hwaccel types * multicast: use C11 threads (#2262) * menu: fix display name for command dialdir (#2251) re_regex changes first pl strings also if it returns an err. Thus in err case we have to clear the pl dname. * account: do not complete dial URI if scheme is included (#2267) * account: do not complete dial URI if scheme is included * test: fix test case for account_uri_complete * menu: simplify URI complete (#2268) * gtk: use new function account_uri_complete_strdup() (#2273) * Removed module avformat dependency on libpostproc (#2274) * make: detect and add swscale module in modules.mk (#2281) * cmake: add APP_MODULES symlinks (#2286) * cmake: add APP_MODULES symlinks * cmake: cleanup mod_lib symlink * docs: update README and supported platforms * cmake: use CMAKE_SHARED_MODULE_SUFFIX (#2292) * version 2.9.0 * fix menu Co-authored-by: juha-h <jh@tutpro.com> Co-authored-by: Maximilian Fridrich <m.fridrich@commend.com> Co-authored-by: Franz <48635806+fAuernigg@users.noreply.github.com> Co-authored-by: Christian Spielberger <c.spielberger@commend.com> Co-authored-by: Sebastian Reimers <hallo@studio-link.de> Co-authored-by: Robert Scheck <robert-scheck@users.noreply.github.com> Co-authored-by: ninp0 <jake.hoopes+github@gmail.com> Co-authored-by: Christoph Huber <c.huber@commend.com> Co-authored-by: Agorgianitis Loukas <agorg_louk@icloud.com>
Sign up for free
to join this conversation on GitHub.
Already have an account?
Sign in to comment
Add this suggestion to a batch that can be applied as a single commit.
This suggestion is invalid because no changes were made to the code.
Suggestions cannot be applied while the pull request is closed.
Suggestions cannot be applied while viewing a subset of changes.
Only one suggestion per line can be applied in a batch.
Add this suggestion to a batch that can be applied as a single commit.
Applying suggestions on deleted lines is not supported.
You must change the existing code in this line in order to create a valid suggestion.
Outdated suggestions cannot be applied.
This suggestion has been applied or marked resolved.
Suggestions cannot be applied from pending reviews.
Suggestions cannot be applied on multi-line comments.
Suggestions cannot be applied while the pull request is queued to merge.
Suggestion cannot be applied right now. Please check back later.
No description provided.