public/rtphone
2
0
Fork 0
Code Issues Pull Requests Projects Releases Wiki Activity

- ongoing work to ease jitter & RTT calculation

Browse Source
This commit is contained in:
Dmytro Bogovych
2023-04-11 13:43:38 +03:00
parent d90940c907
commit 0607bd1c47
14 changed files with 1071 additions and 1037 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
src/engine/helper/HL_StreamState.h
View File

@@ -1,4 +1,4 @@
/* Copyright(C) 2007-2016 VoIP objects (voipobjects.com)
/* Copyright(C) 2007-2023 VoIP objects (voipobjects.com)
* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this
* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
@@ -10,13 +10,13 @@
enum class StreamState
{
Sending = 1, // Transmitting RTP. Set this flag to allow outgoing media stream.
Sending = 1, // Transmitting RTP. Set this flag to allow outgoing media stream.
Receiving = 2, // Receiving RTP. Set this flag to allow receiving media stream.
Playing = 4, // Play to audio. Unmutes the audio from specified stream.
Grabbing = 8, // Capture audio. Unmutes the audio to specified stream.
Srtp = 16, // Use SRTP. Make attempt
SipSend = 32, // Declare send capability in SDP
SipRecv = 64 // Declare recv capability in SDP
Playing = 4, // Play to audio. Unmutes the audio from specified stream.
Grabbing = 8, // Capture audio. Unmutes the audio to specified stream.
Srtp = 16, // Use SRTP. Make attempt
SipSend = 32, // Declare send capability in SDP
SipRecv = 64 // Declare recv capability in SDP
};