merge #9
Closed
merge #9
Conversation
This file contains hidden or 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
…ection, Video stream are added/removed Also add missing code in SendRtpRaw() and in SendRtcpTWCCFeedback() when text as MediaType is used Remove redundancy code Use same tests for video than ones used for audio or text
…structor returns. Can cause exceptions that can't be caught and crash the running app. (sipsorcery-org#1439)
Fix sipsorcery-org#1438: Video Stream are cut when, in a same PeeConnection, Video streams are added/removed
* wip: add turn option to FFmpeg example. * Adjusted webrtc ffmpeg example to optioanlly use a TURN server.
…ipsorcery-org#1446) * Improvements to the webrtc ffmpeg demo ui to make more responsive. * Update examples/WebRTCExamples/WebRTCFFmpegGetStarted/wwwroot/index.html Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/WebRTCExamples/WebRTCFFmpegGetStarted/wwwroot/index.html Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/WebRTCExamples/WebRTCFFmpegGetStarted/wwwroot/index.html Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> --------- Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Removed confusing license wording.
Removed confusing license wording
* Add network change detection to clear cached addresses. * Return cached ip only when it is valid
* wip: adding turn client for rtp session. * wip: wiring up turn client state machine. * Progressed to allocate error response from TURN server. * wip * wip getting turn client working with rtpsession. * wip * Turn client wiring approach working with SIP UAC. * Tweaked logging. * Add max allocate attempts to turn client. * Moved stun message handling into base RTPChannel class. * wip: turn client timers. * Improvements to TURN client timers. Still not quite right. * Turn refresh timers now working properly. * Fixed SO. * Fixed unit tests. * Added UseTurn extension method. * Adjustments to get TURN working for SIP UAC and UAS. * Removed incorrect obsolete attribute on RtpBindAddress. * Fixed typo and adjusted precedence of relay endpoint use in SDP. * Update src/net/TURN/TurnRelayEndPoint.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update src/net/TURN/TurnClientExtensions.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update src/net/TURN/IceTcpReceiver.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/sipcmdline/Program.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/SIPExamples/UserAgentServerWithStun/Program.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/SIPExamples/UserAgentClientWithStun/Program.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update src/net/TURN/TurnRelayEndPoint.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Fixed typos. * Fixed typos. --------- Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
…exive address more convenient (sipsorcery-org#1460) * Adds a STUNClient and extension to make consuming the RTP server reflexive address more convenient. * Update src/net/RTP/RTPSession.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update src/net/STUN/STUNClient.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Use the timeout secodns in the stun client extension. --------- Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
…ipsorcery-org#1461) * wip. * Adjusted TURN client not to rely on SDP connection IP address. Updated examples. * Update examples/SIPExamples/UserAgentClient/Program.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> * Update examples/SIPExamples/UserAgentServer/Program.cs Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com> --------- Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
If we have multiple videotracks in one session, there are multiple ssrc's in the REMB feedback. prior to this fix, there was a warning about unparseable packets as the remeining buffer belongs to the remb feedback.
…r ssrc 0, (sipsorcery-org#1472) * Update RTPSession.cs if we have multiple SSRC in one stream, the REMB feedback comes in for ssrc 0, the real Feedback-SSRCs are in the FeedbackSSRCs array and need to be matched by user. * Update RTPSession.cs use Pattern matching
* Fixing ToUnixTime tests not using DateTimeKind.Utc * Fixing incorrect log in SIPTransactionEngine
Add GetMediaStreamByRTPPort method to enable media stream identification by RTP port when SSRC and payload type matching fails. This provides an additional fallback mechanism in OnReceiveRTPPacket to correctly route incoming RTP packets to their corresponding media streams.
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.