Fix variadic warning for re_printf #4
Closed
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
* srtp: reset srtp_rx for changed remote crypto info * srtp: transmission rekeying command via callid * stream: api to remove media encrytion media object for rekeying * srtp: rekeying testcase * srtp: add missing header * srtp: rework formating * srtp: remove srtp rekeying command + move tx rekeying code into test case + rename srtp_remove_menc_media to srtp_remove_menc_media_state + allow menc_media_state deref only for SRTP module * srtp: implement tx rekeying inside testcase * stream: move menc media state removal function into core.h * srtp: fix function argument list (formatting)
- Adds function audio_update() similar to video_update() that will handle media direction changes, enable/disable audio streams - audio_update() should respect sdp_media disabled flag like video_update().
According to RFC 3261 section 8.1.1.7 the branch parameter in the Via header MUST start with the magic cookie "z9hG4bK". This is now enforced. Some UAs don't comply with this and send multiple requests with the same branch values which can cause issues.
- disable stream before video source is stopped - enable stream after video source was started
…ip#3012) * srtp: lock possible re-keying against usage in receive handler * while the main thread may remove the st->srtp_rx struct, the rx thread accessed the freed memory. * concurrent access of st->srtp_rx is mitigated with lock. * critical section is between first possible removal of st->srtp_rx, until new st->srtp_rx is ready. for the rx thread the access to srtcp_decrypt and srtp_decrypt. * menc: define rekeying handler function for media encryption modules * srtp: locking of srtp_tx and rx objects * critical objects are srtp_tx and srtp_rx, both got there own mutex * any access on these objects are locked with the corresponding lock * receiv and send handler use trylock, therefore are nonblocking * media_txrekey handler defined to generate a new tx key without removing the hole menc_media of the stream * srtp: enable test_srtp_tx_rekey testcase * srtp: avoid deadlock in case srtp_rx == NULL and mtx_rx is not released * srtp: fix possible datarace on st->got_sdp bool. make it atomic
* call,audio: remove audio start/stop redundancy - Function `audio_update()` sets filters, decoder, encoder and starts the audio source. - Function `audio_decoder_set()` sets the decoder and (re-)starts the audio player. - Similar code was removed from call.c. * audio: enable/disable stream at common point
* docs,core: remove reference to multicast * docs: remove ref to multicast
RFC 2976 section 2.2 states that INFO requests with no message body MUST be answered with 200 OK.
Fixes call termination if there are no common audio codecs, but video is possible. In this case now the audio stream is stopped and the call proceeds with the video stream. In bound DTMF is not possible.
A c line with address 0.0.0.0 is allowed. In this case this should not be used as laddr.
Enc/Dec need to be setup first.
* Emit `UA_EVENT_CALL_REMOTE_SDP` if offer in initial INVITE * Emit `UA_EVENT_CALL_LOCAL_SDP` in `sipsess_offer_handler` when local SDP is generated * Do not emit `CALL_EVENT_ANSWERED` in `sipsess_answer_handler` if call is already established
Retrieving the duration of an audio file may lead to long blocking delay. E.g. gst_element_query_duration() Now a callback handler can be called by the application only if needed.
.duration was not initialized
The main thread allocates and sets the filename and emits the module event.
* jack: add jack_server_name config * fix JackServerName option
* ua,menu: move DnD to menu * uag: add uag_reject() with log info * Revert "uag: add uag_reject() with log info" This reverts commit 3c49eb3. * menu: add log info for DnD
* call: add a getter for local stream directions * call: respect local media direction on incoming call
The media direction was never updated if the early audio limit was reached and the call has no video stream.
ENODATA is never returned by `pass_pt_work();`
* added main window for gtk * changed default icon * linter fix * fix linter * notifications for wayland * notifications for wayland * show in all desktop environments * updated desktop file to known working version
…aresip#3173) The warnings are expected because some TLS handshakes in the test are supposed to fail.
…resip#3177) * webrtc_aecm: added missing .le iniitializer * Added also .enabled initializer * Replaced code fix by -Wno-missing-field-initializers Removed .le init from webrtc_aec module
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.