From 1311cd777bc0fcecb8402599da28f98189b31ab2 Mon Sep 17 00:00:00 2001 From: Heikki Tampio Date: Mon, 20 Mar 2023 13:40:51 +0200 Subject: [PATCH] Release 2.3.0 --- cmake/ProjectDetails.cmake | 2 +- docs/Doxyfile | 2 +- docs/README.md | 2 +- docs/html/annotated.html | 2 +- docs/html/classes.html | 2 +- docs/html/classuvgrtp_1_1context-members.html | 2 +- docs/html/classuvgrtp_1_1context.html | 8 +- .../classuvgrtp_1_1media__stream-members.html | 25 +- docs/html/classuvgrtp_1_1media__stream.html | 7 +- docs/html/classuvgrtp_1_1rtcp-members.html | 2 +- docs/html/classuvgrtp_1_1rtcp.html | 4 +- docs/html/classuvgrtp_1_1session-members.html | 2 +- docs/html/classuvgrtp_1_1session.html | 2 +- docs/html/clock_8hh_source.html | 2 +- docs/html/context_8hh_source.html | 2 +- .../dir_9cba3ccb450b89ac6f0c2a18abcbadf0.html | 2 +- .../dir_d44c64559bbebec7f509842c48db8b23.html | 2 +- .../dir_e3f96ff476f8286b6e150580a2f7bd74.html | 2 +- docs/html/files.html | 2 +- docs/html/frame_8hh_source.html | 2 +- docs/html/functions.html | 2 +- docs/html/functions_func.html | 2 +- docs/html/functions_vars.html | 2 +- docs/html/globals.html | 2 +- docs/html/globals_enum.html | 2 +- docs/html/globals_eval.html | 2 +- docs/html/index.html | 2 +- docs/html/lib_8hh_source.html | 2 +- docs/html/media__stream_8hh_source.html | 299 +++--- docs/html/rtcp_8hh_source.html | 947 +++++++++--------- docs/html/session_8hh_source.html | 4 +- ...1_1frame_1_1rtcp__app__packet-members.html | 2 +- ...tuvgrtp_1_1frame_1_1rtcp__app__packet.html | 2 +- ...grtp_1_1frame_1_1rtcp__header-members.html | 2 +- ...structuvgrtp_1_1frame_1_1rtcp__header.html | 2 +- ...ame_1_1rtcp__receiver__report-members.html | 2 +- ...tp_1_1frame_1_1rtcp__receiver__report.html | 2 +- ...1frame_1_1rtcp__report__block-members.html | 2 +- ...vgrtp_1_1frame_1_1rtcp__report__block.html | 2 +- ...1_1frame_1_1rtcp__sdes__chunk-members.html | 2 +- ...tuvgrtp_1_1frame_1_1rtcp__sdes__chunk.html | 2 +- ..._1_1frame_1_1rtcp__sdes__item-members.html | 2 +- ...ctuvgrtp_1_1frame_1_1rtcp__sdes__item.html | 2 +- ..._1frame_1_1rtcp__sdes__packet-members.html | 2 +- ...uvgrtp_1_1frame_1_1rtcp__sdes__packet.html | 2 +- ..._1frame_1_1rtcp__sender__info-members.html | 2 +- ...uvgrtp_1_1frame_1_1rtcp__sender__info.html | 2 +- ...frame_1_1rtcp__sender__report-members.html | 2 +- ...grtp_1_1frame_1_1rtcp__sender__report.html | 2 +- ...uvgrtp_1_1frame_1_1rtp__frame-members.html | 2 +- .../structuvgrtp_1_1frame_1_1rtp__frame.html | 2 +- docs/html/util_8hh.html | 4 +- docs/html/util_8hh_source.html | 2 +- docs/html/version_8hh_source.html | 2 +- docs/html/wrapper__c_8hh_source.html | 2 +- src/media_stream.cc | 3 +- 56 files changed, 706 insertions(+), 689 deletions(-) diff --git a/cmake/ProjectDetails.cmake b/cmake/ProjectDetails.cmake index 70b03761..941263bc 100644 --- a/cmake/ProjectDetails.cmake +++ b/cmake/ProjectDetails.cmake @@ -1,3 +1,3 @@ -set(uvgrtp_VER 2.2.0) +set(uvgrtp_VER 2.3.0) set(uvgrtp_DESCR "uvgRTP is an Real-Time Transport Protocol (RTP) library written in C++ with a focus on simple to use and high-efficiency media delivery over the internet") set(uvgrtp_URL "https://github.com/ultravideo/uvgRTP") \ No newline at end of file diff --git a/docs/Doxyfile b/docs/Doxyfile index 763da42d..0fdc4c48 100644 --- a/docs/Doxyfile +++ b/docs/Doxyfile @@ -48,7 +48,7 @@ PROJECT_NAME = uvgRTP # could be handy for archiving the generated documentation or if some version # control system is used. -PROJECT_NUMBER = 2.2.0 +PROJECT_NUMBER = 2.3.0 # Using the PROJECT_BRIEF tag one can provide an optional one line description # for a project that appears at the top of each page and should give viewer a diff --git a/docs/README.md b/docs/README.md index 8b8270f4..a3383936 100644 --- a/docs/README.md +++ b/docs/README.md @@ -155,7 +155,7 @@ The second way of handling key-management of SRTP is to do it outside uvgRTP. To ## MTU size -The default MTU size of uvgRTP has been set to 1492 to account for 8 bytes of unknown overhead. uvgRTP assumes the presence of an UDP header and IP header in addition an RTP header which are taken into account when fragmenting frames. If your application is expected to work through tunneling such as VPN or IP tunneling which adds additional headers on top of packets, you may need to lower the MTU size to avoid IP level fragmentation. Some networks also allow for a higher MTU size in which case you can increase this. +The default MTU size of uvgRTP has been set to 1492 to account for 8 bytes of unknown overhead. uvgRTP assumes the presence of an UDP header and IP header in addition an RTP header which are taken into account when fragmenting frames. If your application is expected to work through tunneling such as VPN or IPv6 to IPv4 which add additional headers on top of packets, you may need to lower the MTU size to avoid IP level fragmentation. Some networks also allow for a higher MTU size in which case you can increase this. ## uvgRTP video reception behavior with packet loss diff --git a/docs/html/annotated.html b/docs/html/annotated.html index 64d1c913..ebd4fcfa 100644 --- a/docs/html/annotated.html +++ b/docs/html/annotated.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/classes.html b/docs/html/classes.html index c32d2b1e..3f890f70 100644 --- a/docs/html/classes.html +++ b/docs/html/classes.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/classuvgrtp_1_1context-members.html b/docs/html/classuvgrtp_1_1context-members.html index c4f73b27..c05f78e2 100644 --- a/docs/html/classuvgrtp_1_1context-members.html +++ b/docs/html/classuvgrtp_1_1context-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/classuvgrtp_1_1context.html b/docs/html/classuvgrtp_1_1context.html index c29d3782..5e5127bc 100644 --- a/docs/html/classuvgrtp_1_1context.html +++ b/docs/html/classuvgrtp_1_1context.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -172,7 +172,7 @@

Parameters
- +
addressIPv4 address of the remote participant
addressIP address of the remote participant
@@ -217,8 +217,8 @@

RCE_HOLEPUNCH_KEEPALIVE

Parameters
- - + +
remote_addrIPv4 address of the remote participant
local_addrIPv4 address of a local interface
remote_addrIP address of the remote participant
local_addrIP address of a local interface
diff --git a/docs/html/classuvgrtp_1_1media__stream-members.html b/docs/html/classuvgrtp_1_1media__stream-members.html index 42f3d9a6..5b134c9e 100644 --- a/docs/html/classuvgrtp_1_1media__stream-members.html +++ b/docs/html/classuvgrtp_1_1media__stream-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -82,17 +82,18 @@ - - - - - - - - - - - + + + + + + + + + + + +
add_srtp_ctx(uint8_t *key, uint8_t *salt)uvgrtp::media_stream
configure_ctx(int rcc_flag, ssize_t value)uvgrtp::media_stream
get_rtcp()uvgrtp::media_stream
get_ssrc() constuvgrtp::media_stream
install_receive_hook(void *arg, void(*hook)(void *, uvgrtp::frame::rtp_frame *))uvgrtp::media_stream
pull_frame()uvgrtp::media_stream
pull_frame(size_t timeout_ms)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, int rtp_flags)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, uint32_t ts, int rtp_flags)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, uint32_t ts, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags)uvgrtp::media_stream
get_ipv6() const (defined in uvgrtp::media_stream)uvgrtp::media_stream
get_rtcp()uvgrtp::media_stream
get_ssrc() constuvgrtp::media_stream
install_receive_hook(void *arg, void(*hook)(void *, uvgrtp::frame::rtp_frame *))uvgrtp::media_stream
pull_frame()uvgrtp::media_stream
pull_frame(size_t timeout_ms)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, int rtp_flags)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, uint32_t ts, int rtp_flags)uvgrtp::media_stream
push_frame(uint8_t *data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, uint32_t ts, int rtp_flags)uvgrtp::media_stream
push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags)uvgrtp::media_stream
diff --git a/docs/html/classuvgrtp_1_1media__stream.html b/docs/html/classuvgrtp_1_1media__stream.html index 56ae7d9d..3b60be8d 100644 --- a/docs/html/classuvgrtp_1_1media__stream.html +++ b/docs/html/classuvgrtp_1_1media__stream.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -127,6 +127,9 @@ uint32_t get_ssrc () const  Get SSRC identifier. You can use the SSRC value for example to find the report block belonging to this media_stream in RTCP sender/receiver report.
  + +bool get_ipv6 () const + 

Detailed Description

The media_stream is an entity which represents one RTP stream.

@@ -134,7 +137,7 @@

If RCE_RTCP was given when creating media_stream, you can get the uvgrtp::rtcp object with get_rtcp()-function.

media_stream corresponds to one RTP session in RFC 3550.

-

Definition at line 49 of file media_stream.hh.

+

Definition at line 50 of file media_stream.hh.

Member Function Documentation

◆ add_srtp_ctx()

diff --git a/docs/html/classuvgrtp_1_1rtcp-members.html b/docs/html/classuvgrtp_1_1rtcp-members.html index b149d59c..d1d061b8 100644 --- a/docs/html/classuvgrtp_1_1rtcp-members.html +++ b/docs/html/classuvgrtp_1_1rtcp-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/classuvgrtp_1_1rtcp.html b/docs/html/classuvgrtp_1_1rtcp.html index 43a427f7..6a8ca313 100644 --- a/docs/html/classuvgrtp_1_1rtcp.html +++ b/docs/html/classuvgrtp_1_1rtcp.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -141,7 +141,7 @@

You can use the APP packet to test new RTCP packet types using the send_app_packet()-function. The APP packets are added to these periodically sent compound packets.

See RFC 3550 section 6 for more details.

-

Definition at line 111 of file rtcp.hh.

+

Definition at line 118 of file rtcp.hh.

Member Function Documentation

◆ install_app_hook() [1/2]

diff --git a/docs/html/classuvgrtp_1_1session-members.html b/docs/html/classuvgrtp_1_1session-members.html index 98ba13ae..5bdcce61 100644 --- a/docs/html/classuvgrtp_1_1session-members.html +++ b/docs/html/classuvgrtp_1_1session-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/classuvgrtp_1_1session.html b/docs/html/classuvgrtp_1_1session.html index 98f838cf..8599b6ef 100644 --- a/docs/html/classuvgrtp_1_1session.html +++ b/docs/html/classuvgrtp_1_1session.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/clock_8hh_source.html b/docs/html/clock_8hh_source.html index 3726cd27..06071677 100644 --- a/docs/html/clock_8hh_source.html +++ b/docs/html/clock_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/context_8hh_source.html b/docs/html/context_8hh_source.html index 416c23de..1813db08 100644 --- a/docs/html/context_8hh_source.html +++ b/docs/html/context_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/dir_9cba3ccb450b89ac6f0c2a18abcbadf0.html b/docs/html/dir_9cba3ccb450b89ac6f0c2a18abcbadf0.html index aeeeb215..6ddf74d7 100644 --- a/docs/html/dir_9cba3ccb450b89ac6f0c2a18abcbadf0.html +++ b/docs/html/dir_9cba3ccb450b89ac6f0c2a18abcbadf0.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/dir_d44c64559bbebec7f509842c48db8b23.html b/docs/html/dir_d44c64559bbebec7f509842c48db8b23.html index cdad309c..86b00314 100644 --- a/docs/html/dir_d44c64559bbebec7f509842c48db8b23.html +++ b/docs/html/dir_d44c64559bbebec7f509842c48db8b23.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/dir_e3f96ff476f8286b6e150580a2f7bd74.html b/docs/html/dir_e3f96ff476f8286b6e150580a2f7bd74.html index acd2aa89..9f5b9a09 100644 --- a/docs/html/dir_e3f96ff476f8286b6e150580a2f7bd74.html +++ b/docs/html/dir_e3f96ff476f8286b6e150580a2f7bd74.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/files.html b/docs/html/files.html index 619ba91b..8f24c3bd 100644 --- a/docs/html/files.html +++ b/docs/html/files.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/frame_8hh_source.html b/docs/html/frame_8hh_source.html index afa7cadf..22a749ab 100644 --- a/docs/html/frame_8hh_source.html +++ b/docs/html/frame_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/functions.html b/docs/html/functions.html index f4222fbe..84df2fce 100644 --- a/docs/html/functions.html +++ b/docs/html/functions.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/functions_func.html b/docs/html/functions_func.html index 9aa214c3..5080e775 100644 --- a/docs/html/functions_func.html +++ b/docs/html/functions_func.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/functions_vars.html b/docs/html/functions_vars.html index 19e16b2b..6e09b510 100644 --- a/docs/html/functions_vars.html +++ b/docs/html/functions_vars.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/globals.html b/docs/html/globals.html index 602226d6..19c1ce18 100644 --- a/docs/html/globals.html +++ b/docs/html/globals.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/globals_enum.html b/docs/html/globals_enum.html index fe3b7c1d..8610f3ed 100644 --- a/docs/html/globals_enum.html +++ b/docs/html/globals_enum.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/globals_eval.html b/docs/html/globals_eval.html index ba406323..3d04d685 100644 --- a/docs/html/globals_eval.html +++ b/docs/html/globals_eval.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/index.html b/docs/html/index.html index 85f7f728..e9370002 100644 --- a/docs/html/index.html +++ b/docs/html/index.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/lib_8hh_source.html b/docs/html/lib_8hh_source.html index 49e6c96d..2edd0b73 100644 --- a/docs/html/lib_8hh_source.html +++ b/docs/html/lib_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/media__stream_8hh_source.html b/docs/html/media__stream_8hh_source.html index 7f99d866..0ea576a4 100644 --- a/docs/html/media__stream_8hh_source.html +++ b/docs/html/media__stream_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -89,157 +89,160 @@
10#ifndef _WIN32
11#include <sys/socket.h>
12#include <netinet/in.h>
-
13#endif
-
14
-
15namespace uvgrtp {
-
16
-
17 // forward declarations
-
18 class rtp;
-
19 class rtcp;
-
20
-
21 class zrtp;
-
22 class base_srtp;
-
23 class srtp;
-
24 class srtcp;
-
25
-
26 class reception_flow;
-
27 class holepuncher;
-
28 class socket;
-
29
-
30 namespace frame {
-
31 struct rtp_frame;
-
32 }
-
33
-
34 namespace formats {
-
35 class media;
-
36 }
-
37
-
49 class media_stream {
-
50 public:
-
52 media_stream(std::string cname, std::string remote_addr, std::string local_addr, uint16_t src_port, uint16_t dst_port,
-
53 rtp_format_t fmt, int rce_flags);
-
54 ~media_stream();
-
55
-
56 /* Initialize traditional RTP session
-
57 * Allocate Connection/Reader/Writer objects and initialize them
-
58 *
-
59 * Return RTP_OK on success
-
60 * Return RTP_MEMORY_ERROR if allocation failed
-
61 *
-
62 * Other error return codes are defined in {conn,writer,reader}.hh */
-
63 rtp_error_t init();
-
64
-
65 /* Initialize Secure RTP session
-
66 * Allocate Connection/Reader/Writer objects and initialize them
-
67 *
-
68 * Return RTP_OK on success
-
69 * Return RTP_MEMORY_ERROR if allocation failed
-
70 *
-
71 * TODO document all error codes!
-
72 *
-
73 * Other error return codes are defined in {conn,writer,reader,srtp}.hh */
-
74 rtp_error_t init(std::shared_ptr<uvgrtp::zrtp> zrtp);
-
76
-
93 rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt);
-
94
-
117 rtp_error_t push_frame(uint8_t *data, size_t data_len, int rtp_flags);
-
118
-
141 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, int rtp_flags);
-
142
-
172 rtp_error_t push_frame(uint8_t *data, size_t data_len, uint32_t ts, int rtp_flags);
-
173
-
206 rtp_error_t push_frame(uint8_t* data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags);
-
207
-
237 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, int rtp_flags);
-
238
-
271 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags);
-
272
-
281 uvgrtp::frame::rtp_frame *pull_frame();
-
282
-
293 uvgrtp::frame::rtp_frame *pull_frame(size_t timeout_ms);
-
294
-
313 rtp_error_t install_receive_hook(void *arg, void (*hook)(void *, uvgrtp::frame::rtp_frame *));
-
314
-
324 rtp_error_t configure_ctx(int rcc_flag, ssize_t value);
-
325
-
327
-
328 /* Get unique key of the media stream
-
329 * Used by session to index media streams */
-
330 uint32_t get_key() const;
-
332
-
344 uvgrtp::rtcp *get_rtcp();
-
345
-
352 uint32_t get_ssrc() const;
-
353
-
354 private:
-
355 /* Initialize the connection by initializing the socket
-
356 * and binding ourselves to specified interface and creating
-
357 * an outgoing address */
-
358 rtp_error_t init_connection();
-
359
-
360 /* Create the media object for the stream */
-
361 rtp_error_t create_media(rtp_format_t fmt);
-
362
-
363 /* free all allocated resources */
-
364 rtp_error_t free_resources(rtp_error_t ret);
-
365
-
366 rtp_error_t init_srtp_with_zrtp(int rce_flags, int type, std::shared_ptr<uvgrtp::base_srtp> srtp,
-
367 std::shared_ptr<uvgrtp::zrtp> zrtp);
-
368
-
369 rtp_error_t start_components();
-
370
-
371 uint32_t get_default_bandwidth_kbps(rtp_format_t fmt);
-
372
-
373 bool check_pull_preconditions();
-
374 rtp_error_t check_push_preconditions(int rtp_flags, bool smart_pointer);
-
375
-
376 inline uint8_t* copy_frame(uint8_t* original, size_t data_len);
-
377
-
378 uint32_t key_;
-
379
-
380 std::shared_ptr<uvgrtp::srtp> srtp_;
-
381 std::shared_ptr<uvgrtp::srtcp> srtcp_;
-
382 std::shared_ptr<uvgrtp::socket> socket_;
-
383 std::shared_ptr<uvgrtp::rtp> rtp_;
-
384 std::shared_ptr<uvgrtp::rtcp> rtcp_;
-
385
-
386 sockaddr_in remote_sockaddr_;
-
387 std::string remote_address_;
-
388 std::string local_address_;
-
389 uint16_t src_port_;
-
390 uint16_t dst_port_;
-
391 rtp_format_t fmt_;
-
392
-
393 /* Media context config */
-
394 int rce_flags_ = 0;
+
13#else
+
14#include <ws2ipdef.h>
+
15#endif
+
16namespace uvgrtp {
+
17
+
18 // forward declarations
+
19 class rtp;
+
20 class rtcp;
+
21
+
22 class zrtp;
+
23 class base_srtp;
+
24 class srtp;
+
25 class srtcp;
+
26
+
27 class reception_flow;
+
28 class holepuncher;
+
29 class socket;
+
30
+
31 namespace frame {
+
32 struct rtp_frame;
+
33 }
+
34
+
35 namespace formats {
+
36 class media;
+
37 }
+
38
+
50 class media_stream {
+
51 public:
+
53 media_stream(std::string cname, std::string remote_addr, std::string local_addr, uint16_t src_port, uint16_t dst_port,
+
54 rtp_format_t fmt, int rce_flags);
+
55 ~media_stream();
+
56
+
57 /* Initialize traditional RTP session
+
58 * Allocate Connection/Reader/Writer objects and initialize them
+
59 *
+
60 * Return RTP_OK on success
+
61 * Return RTP_MEMORY_ERROR if allocation failed
+
62 *
+
63 * Other error return codes are defined in {conn,writer,reader}.hh */
+
64 rtp_error_t init();
+
65
+
66 /* Initialize Secure RTP session
+
67 * Allocate Connection/Reader/Writer objects and initialize them
+
68 *
+
69 * Return RTP_OK on success
+
70 * Return RTP_MEMORY_ERROR if allocation failed
+
71 *
+
72 * TODO document all error codes!
+
73 *
+
74 * Other error return codes are defined in {conn,writer,reader,srtp}.hh */
+
75 rtp_error_t init(std::shared_ptr<uvgrtp::zrtp> zrtp);
+
77
+
94 rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt);
+
95
+
118 rtp_error_t push_frame(uint8_t *data, size_t data_len, int rtp_flags);
+
119
+
142 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, int rtp_flags);
+
143
+
173 rtp_error_t push_frame(uint8_t *data, size_t data_len, uint32_t ts, int rtp_flags);
+
174
+
207 rtp_error_t push_frame(uint8_t* data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags);
+
208
+
238 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, int rtp_flags);
+
239
+
272 rtp_error_t push_frame(std::unique_ptr<uint8_t[]> data, size_t data_len, uint32_t ts, uint64_t ntp_ts, int rtp_flags);
+
273
+
282 uvgrtp::frame::rtp_frame *pull_frame();
+
283
+
294 uvgrtp::frame::rtp_frame *pull_frame(size_t timeout_ms);
+
295
+
314 rtp_error_t install_receive_hook(void *arg, void (*hook)(void *, uvgrtp::frame::rtp_frame *));
+
315
+
325 rtp_error_t configure_ctx(int rcc_flag, ssize_t value);
+
326
+
328
+
329 /* Get unique key of the media stream
+
330 * Used by session to index media streams */
+
331 uint32_t get_key() const;
+
333
+
345 uvgrtp::rtcp *get_rtcp();
+
346
+
353 uint32_t get_ssrc() const;
+
354
+
355 private:
+
356 /* Initialize the connection by initializing the socket
+
357 * and binding ourselves to specified interface and creating
+
358 * an outgoing address */
+
359 rtp_error_t init_connection();
+
360
+
361 /* Create the media object for the stream */
+
362 rtp_error_t create_media(rtp_format_t fmt);
+
363
+
364 /* free all allocated resources */
+
365 rtp_error_t free_resources(rtp_error_t ret);
+
366
+
367 rtp_error_t init_srtp_with_zrtp(int rce_flags, int type, std::shared_ptr<uvgrtp::base_srtp> srtp,
+
368 std::shared_ptr<uvgrtp::zrtp> zrtp);
+
369
+
370 rtp_error_t start_components();
+
371
+
372 uint32_t get_default_bandwidth_kbps(rtp_format_t fmt);
+
373
+
374 bool check_pull_preconditions();
+
375 rtp_error_t check_push_preconditions(int rtp_flags, bool smart_pointer);
+
376
+
377 inline uint8_t* copy_frame(uint8_t* original, size_t data_len);
+
378
+
379 uint32_t key_;
+
380
+
381 std::shared_ptr<uvgrtp::srtp> srtp_;
+
382 std::shared_ptr<uvgrtp::srtcp> srtcp_;
+
383 std::shared_ptr<uvgrtp::socket> socket_;
+
384 std::shared_ptr<uvgrtp::rtp> rtp_;
+
385 std::shared_ptr<uvgrtp::rtcp> rtcp_;
+
386
+
387 sockaddr_in remote_sockaddr_;
+
388 sockaddr_in6 remote_sockaddr_ip6_;
+
389 std::string remote_address_;
+
390 std::string local_address_;
+
391 uint16_t src_port_;
+
392 uint16_t dst_port_;
+
393 bool ipv6_;
+
394 rtp_format_t fmt_;
395
-
396 /* Has the media stream been initialized */
-
397 bool initialized_;
+
396 /* Media context config */
+
397 int rce_flags_ = 0;
398
-
399 /* Primary handler keys for the RTP reception flow */
-
400 uint32_t rtp_handler_key_;
-
401 uint32_t zrtp_handler_key_;
-
402
-
403 /* RTP packet reception flow. Dispatches packets to other components */
-
404 std::unique_ptr<uvgrtp::reception_flow> reception_flow_;
+
399 /* Has the media stream been initialized */
+
400 bool initialized_;
+
401
+
402 /* Primary handler keys for the RTP reception flow */
+
403 uint32_t rtp_handler_key_;
+
404 uint32_t zrtp_handler_key_;
405
-
406 /* Media object associated with this media stream. */
-
407 std::unique_ptr<uvgrtp::formats::media> media_;
+
406 /* RTP packet reception flow. Dispatches packets to other components */
+
407 std::unique_ptr<uvgrtp::reception_flow> reception_flow_;
408
-
409 /* Thread that keeps the holepunched connection open for unidirectional streams */
-
410 std::unique_ptr<uvgrtp::holepuncher> holepuncher_;
+
409 /* Media object associated with this media stream. */
+
410 std::unique_ptr<uvgrtp::formats::media> media_;
411
-
412 std::string cname_;
-
413
-
414 ssize_t fps_numerator_ = 30;
-
415 ssize_t fps_denominator_ = 1;
-
416 uint32_t bandwidth_ = 0;
-
417 std::shared_ptr<std::atomic<std::uint32_t>> ssrc_;
-
418 };
-
419}
-
420
-
421namespace uvg_rtp = uvgrtp;
-
uvgrtp::media_stream
The media_stream is an entity which represents one RTP stream.
Definition: media_stream.hh:49
+
412 /* Thread that keeps the holepunched connection open for unidirectional streams */
+
413 std::unique_ptr<uvgrtp::holepuncher> holepuncher_;
+
414
+
415 std::string cname_;
+
416
+
417 ssize_t fps_numerator_ = 30;
+
418 ssize_t fps_denominator_ = 1;
+
419 uint32_t bandwidth_ = 0;
+
420 std::shared_ptr<std::atomic<std::uint32_t>> ssrc_;
+
421 };
+
422}
+
423
+
424namespace uvg_rtp = uvgrtp;
+
uvgrtp::media_stream
The media_stream is an entity which represents one RTP stream.
Definition: media_stream.hh:50
uvgrtp::media_stream::push_frame
rtp_error_t push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, int rtp_flags)
Send data to remote participant with a custom timestamp.
uvgrtp::media_stream::add_srtp_ctx
rtp_error_t add_srtp_ctx(uint8_t *key, uint8_t *salt)
Add keying information for user-managed SRTP session.
uvgrtp::media_stream::push_frame
rtp_error_t push_frame(uint8_t *data, size_t data_len, int rtp_flags)
Send data to remote participant with a custom timestamp.
@@ -253,7 +256,7 @@
uvgrtp::media_stream::get_rtcp
uvgrtp::rtcp * get_rtcp()
Get pointer to the RTCP object of the media stream.
uvgrtp::media_stream::push_frame
rtp_error_t push_frame(std::unique_ptr< uint8_t[]> data, size_t data_len, uint32_t ts, int rtp_flags)
Send data to remote participant with a custom timestamp.
uvgrtp::media_stream::push_frame
rtp_error_t push_frame(uint8_t *data, size_t data_len, uint32_t ts, int rtp_flags)
Send data to remote participant with a custom timestamp.
-
uvgrtp::rtcp
RTCP instance handles all incoming and outgoing RTCP traffic, including report generation.
Definition: rtcp.hh:111
+
uvgrtp::rtcp
RTCP instance handles all incoming and outgoing RTCP traffic, including report generation.
Definition: rtcp.hh:118
uvgrtp::frame::rtp_frame
See RFC 3550 section 5
Definition: frame.hh:53
diff --git a/docs/html/rtcp_8hh_source.html b/docs/html/rtcp_8hh_source.html index 33f1c7bb..9f2cfdbf 100644 --- a/docs/html/rtcp_8hh_source.html +++ b/docs/html/rtcp_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -83,503 +83,512 @@
4#include "util.hh"
5#include "frame.hh"
6
-
7#include <bitset>
-
8#include <map>
-
9#include <thread>
-
10#include <vector>
-
11#include <functional>
-
12#include <memory>
-
13#include <mutex>
-
14#include <deque>
-
15#include <atomic>
-
16
-
17namespace uvgrtp {
-
18
-
19 class rtp;
-
20 class srtcp;
-
21 class socket;
-
22
-
23 typedef std::vector<std::pair<size_t, uint8_t*>> buf_vec; // also defined in socket.hh
-
24
-
26 enum RTCP_ROLE {
-
27 RECEIVER,
-
28 SENDER
-
29 };
-
30
-
31 struct sender_statistics {
-
32 /* sender stats */
-
33 uint32_t sent_pkts = 0; /* Number of sent RTP packets */
-
34 uint32_t sent_bytes = 0; /* Number of sent bytes excluding RTP Header */
-
35 bool sent_rtp_packet = false; // since last report
+
7#ifdef _WIN32
+
8#include <ws2ipdef.h>
+
9#else
+
10#include <sys/socket.h>
+
11#include <netinet/in.h>
+
12#endif
+
13
+
14#include <bitset>
+
15#include <map>
+
16#include <thread>
+
17#include <vector>
+
18#include <functional>
+
19#include <memory>
+
20#include <mutex>
+
21#include <deque>
+
22#include <atomic>
+
23
+
24namespace uvgrtp {
+
25
+
26 class rtp;
+
27 class srtcp;
+
28 class socket;
+
29
+
30 typedef std::vector<std::pair<size_t, uint8_t*>> buf_vec; // also defined in socket.hh
+
31
+
33 enum RTCP_ROLE {
+
34 RECEIVER,
+
35 SENDER
36 };
37
-
38 struct receiver_statistics {
-
39 /* receiver stats */
-
40 uint32_t received_pkts = 0; /* Number of packets received */
-
41 uint32_t lost_pkts = 0; /* Number of dropped RTP packets */
-
42 uint32_t received_bytes = 0; /* Number of bytes received excluding RTP Header */
-
43 bool received_rtp_packet = false; // since last report
+
38 struct sender_statistics {
+
39 /* sender stats */
+
40 uint32_t sent_pkts = 0; /* Number of sent RTP packets */
+
41 uint32_t sent_bytes = 0; /* Number of sent bytes excluding RTP Header */
+
42 bool sent_rtp_packet = false; // since last report
+
43 };
44
-
45 uint32_t expected_pkts = 0; /* Number of expected packets */
-
46 uint32_t received_prior = 0; /* Number of received packets in last report */
-
47 uint32_t expected_prior = 0; /* Number of expected packets in last report */
-
48
-
49 double jitter = 0; /* The estimation of jitter (see RFC 3550 A.8) */
-
50 uint32_t transit = 0; /* TODO: */
+
45 struct receiver_statistics {
+
46 /* receiver stats */
+
47 uint32_t received_pkts = 0; /* Number of packets received */
+
48 uint32_t lost_pkts = 0; /* Number of dropped RTP packets */
+
49 uint32_t received_bytes = 0; /* Number of bytes received excluding RTP Header */
+
50 bool received_rtp_packet = false; // since last report
51
-
52
-
53 /* Receiver clock related stuff */
-
54 uint64_t initial_ntp = 0; /* Wallclock reading when the first RTP packet was received */
-
55 uint32_t initial_rtp = 0; /* RTP timestamp of the first RTP packet received */
-
56 uint32_t clock_rate = 0; /* Rate of the clock (used for jitter calculations) */
-
57
-
58 uint32_t lsr = 0; /* Middle 32 bits of the 64-bit NTP timestamp of previous SR */
-
59 uvgrtp::clock::hrc::hrc_t sr_ts; /* When the last SR was received (used to calculate delay) */
-
60
-
61 uint16_t max_seq = 0; /* Highest sequence number received */
-
62 uint32_t base_seq = 0; /* First sequence number received */
-
63 uint32_t bad_seq = 0; /* TODO: */
-
64 uint16_t cycles = 0; /* Number of sequence cycles */
-
65 };
-
66
-
67 struct rtcp_participant {
-
68 struct receiver_statistics stats; /* RTCP session statistics of the participant */
-
69
-
70 uint32_t probation = 0; /* has the participant been fully accepted to the session */
-
71 int role = 0; /* is the participant a sender or a receiver */
-
72
-
73 /* Save the latest RTCP packets received from this participant
-
74 * Users can query these packets using the SSRC of participant */
-
75 uvgrtp::frame::rtcp_sender_report *sr_frame = nullptr;
-
76 uvgrtp::frame::rtcp_receiver_report *rr_frame = nullptr;
-
77 uvgrtp::frame::rtcp_sdes_packet *sdes_frame = nullptr;
-
78 uvgrtp::frame::rtcp_app_packet *app_frame = nullptr;
-
79 };
-
80
-
81 struct rtcp_app_packet {
-
82 rtcp_app_packet(const rtcp_app_packet& orig_packet) = delete;
-
83 rtcp_app_packet(const char* name, uint8_t subtype, uint32_t payload_len, std::unique_ptr<uint8_t[]> payload);
-
84 ~rtcp_app_packet();
-
85
-
86 const char* name;
-
87 uint8_t subtype;
-
88
-
89 uint32_t payload_len;
-
90 std::unique_ptr<uint8_t[]> payload;
-
91 };
-
93
-
111 class rtcp {
-
112 public:
-
114 rtcp(std::shared_ptr<uvgrtp::rtp> rtp, std::shared_ptr<std::atomic<std::uint32_t>> ssrc, std::string cname, int rce_flags);
-
115 rtcp(std::shared_ptr<uvgrtp::rtp> rtp, std::shared_ptr<std::atomic<std::uint32_t>> ssrc, std::string cname, std::shared_ptr<uvgrtp::srtcp> srtcp, int rce_flags);
-
116 ~rtcp();
-
117
-
118 /* start the RTCP runner thread
-
119 *
-
120 * return RTP_OK on success and RTP_MEMORY_ERROR if the allocation fails */
-
121 rtp_error_t start();
-
122
-
123 /* End the RTCP session and send RTCP BYE to all participants
-
124 *
-
125 * return RTP_OK on success */
-
126 rtp_error_t stop();
-
127
-
128 /* Generate either RTCP Sender or Receiver report and sent it to all participants
-
129 * Return RTP_OK on success and RTP_ERROR on error */
-
130 rtp_error_t generate_report();
-
131
-
132 /* Handle incoming RTCP packet (first make sure it's a valid RTCP packet)
-
133 * This function will call one of the above functions internally
-
134 *
-
135 * Return RTP_OK on success and RTP_ERROR on error */
-
136 rtp_error_t handle_incoming_packet(uint8_t *buffer, size_t size);
+
52 uint32_t expected_pkts = 0; /* Number of expected packets */
+
53 uint32_t received_prior = 0; /* Number of received packets in last report */
+
54 uint32_t expected_prior = 0; /* Number of expected packets in last report */
+
55
+
56 double jitter = 0; /* The estimation of jitter (see RFC 3550 A.8) */
+
57 uint32_t transit = 0; /* TODO: */
+
58
+
59
+
60 /* Receiver clock related stuff */
+
61 uint64_t initial_ntp = 0; /* Wallclock reading when the first RTP packet was received */
+
62 uint32_t initial_rtp = 0; /* RTP timestamp of the first RTP packet received */
+
63 uint32_t clock_rate = 0; /* Rate of the clock (used for jitter calculations) */
+
64
+
65 uint32_t lsr = 0; /* Middle 32 bits of the 64-bit NTP timestamp of previous SR */
+
66 uvgrtp::clock::hrc::hrc_t sr_ts; /* When the last SR was received (used to calculate delay) */
+
67
+
68 uint16_t max_seq = 0; /* Highest sequence number received */
+
69 uint32_t base_seq = 0; /* First sequence number received */
+
70 uint32_t bad_seq = 0; /* TODO: */
+
71 uint16_t cycles = 0; /* Number of sequence cycles */
+
72 };
+
73
+
74 struct rtcp_participant {
+
75 struct receiver_statistics stats; /* RTCP session statistics of the participant */
+
76
+
77 uint32_t probation = 0; /* has the participant been fully accepted to the session */
+
78 int role = 0; /* is the participant a sender or a receiver */
+
79
+
80 /* Save the latest RTCP packets received from this participant
+
81 * Users can query these packets using the SSRC of participant */
+
82 uvgrtp::frame::rtcp_sender_report *sr_frame = nullptr;
+
83 uvgrtp::frame::rtcp_receiver_report *rr_frame = nullptr;
+
84 uvgrtp::frame::rtcp_sdes_packet *sdes_frame = nullptr;
+
85 uvgrtp::frame::rtcp_app_packet *app_frame = nullptr;
+
86 };
+
87
+
88 struct rtcp_app_packet {
+
89 rtcp_app_packet(const rtcp_app_packet& orig_packet) = delete;
+
90 rtcp_app_packet(const char* name, uint8_t subtype, uint32_t payload_len, std::unique_ptr<uint8_t[]> payload);
+
91 ~rtcp_app_packet();
+
92
+
93 const char* name;
+
94 uint8_t subtype;
+
95
+
96 uint32_t payload_len;
+
97 std::unique_ptr<uint8_t[]> payload;
+
98 };
+
100
+
118 class rtcp {
+
119 public:
+
121 rtcp(std::shared_ptr<uvgrtp::rtp> rtp, std::shared_ptr<std::atomic<std::uint32_t>> ssrc, std::string cname, int rce_flags);
+
122 rtcp(std::shared_ptr<uvgrtp::rtp> rtp, std::shared_ptr<std::atomic<std::uint32_t>> ssrc, std::string cname, std::shared_ptr<uvgrtp::srtcp> srtcp, int rce_flags);
+
123 ~rtcp();
+
124
+
125 /* start the RTCP runner thread
+
126 *
+
127 * return RTP_OK on success and RTP_MEMORY_ERROR if the allocation fails */
+
128 rtp_error_t start();
+
129
+
130 /* End the RTCP session and send RTCP BYE to all participants
+
131 *
+
132 * return RTP_OK on success */
+
133 rtp_error_t stop();
+
134
+
135 /* Generate either RTCP Sender or Receiver report and sent it to all participants
+
136 * Return RTP_OK on success and RTP_ERROR on error */
+
137 rtp_error_t generate_report();
138
-
139 /* Send "frame" to all participants
-
140 *
-
141 * These routines will convert all necessary fields to network byte order
-
142 *
-
143 * Return RTP_OK on success
-
144 * Return RTP_INVALID_VALUE if "frame" is in some way invalid
-
145 * Return RTP_SEND_ERROR if sending "frame" did not succeed (see socket.hh for details) */
-
146
-
156 rtp_error_t send_sdes_packet(const std::vector<uvgrtp::frame::rtcp_sdes_item>& items);
-
157
-
170 rtp_error_t send_app_packet(const char *name, uint8_t subtype, uint32_t payload_len, const uint8_t *payload);
-
171
-
186 rtp_error_t send_bye_packet(std::vector<uint32_t> ssrcs);
-
187
-
189 /* Return the latest RTCP packet received from participant of "ssrc"
-
190 * Return nullptr if we haven't received this kind of packet or if "ssrc" doesn't exist
-
191 *
-
192 * NOTE: Caller is responsible for deallocating the memory */
-
193 uvgrtp::frame::rtcp_sender_report *get_sender_packet(uint32_t ssrc);
-
194 uvgrtp::frame::rtcp_receiver_report *get_receiver_packet(uint32_t ssrc);
-
195 uvgrtp::frame::rtcp_sdes_packet *get_sdes_packet(uint32_t ssrc);
-
196 uvgrtp::frame::rtcp_app_packet *get_app_packet(uint32_t ssrc);
-
197
-
198 /* Somebody joined the multicast group the owner of this RTCP instance is part of
-
199 * Add it to RTCP participant list so we can start listening for reports
-
200 *
-
201 * "clock_rate" tells how much the RTP timestamp advances, this information is needed
-
202 * to calculate the interarrival jitter correctly. It has nothing do with our clock rate,
-
203 * (or whether we're even sending anything)
-
204 *
-
205 * Return RTP_OK on success and RTP_ERROR on error */
-
206 rtp_error_t add_initial_participant(uint32_t clock_rate);
-
207
-
208 /* Functions for updating various RTP sender statistics */
-
209 void sender_update_stats(const uvgrtp::frame::rtp_frame *frame);
-
210
-
211 /* If we've detected that our SSRC has collided with someone else's SSRC, we need to
-
212 * generate new random SSRC and reinitialize our own RTCP state.
-
213 * RTCP object still has the participants of "last session", we can use their SSRCs
-
214 * to detected new collision
-
215 *
-
216 * Return RTP_OK if reinitialization succeeded
-
217 * Return RTP_SSRC_COLLISION if our new SSRC has collided and we need to generate new SSRC */
-
218 rtp_error_t reset_rtcp_state(uint32_t ssrc);
-
219
-
220 /* Update various session statistics */
-
221 void update_session_statistics(const uvgrtp::frame::rtp_frame *frame);
-
222
-
223 /* Getter for interval_ms_, which is calculated by set_session_bandwidth */
-
224 uint32_t get_rtcp_interval_ms() const;
-
225
-
226 /* Set RTCP packet transmission interval in milliseconds
-
227 *
-
228 * Return RTP_OK if interval was set successfully
-
229 * Return RTP_INVALID_VALUE if new interval is invalid */
-
230 rtp_error_t set_rtcp_interval_ms(int32_t new_interval);
-
231
-
232 /* Set total bandwidth for this session, called at the start
-
233 * If you want to set the interval manually later, use
-
234 * set_rtcp_interval_ms() function */
-
235 void set_session_bandwidth(uint32_t kbps);
-
236
-
237 std::shared_ptr<uvgrtp::socket> get_socket() const;
+
139 /* Handle incoming RTCP packet (first make sure it's a valid RTCP packet)
+
140 * This function will call one of the above functions internally
+
141 *
+
142 * Return RTP_OK on success and RTP_ERROR on error */
+
143 rtp_error_t handle_incoming_packet(uint8_t *buffer, size_t size);
+
145
+
146 /* Send "frame" to all participants
+
147 *
+
148 * These routines will convert all necessary fields to network byte order
+
149 *
+
150 * Return RTP_OK on success
+
151 * Return RTP_INVALID_VALUE if "frame" is in some way invalid
+
152 * Return RTP_SEND_ERROR if sending "frame" did not succeed (see socket.hh for details) */
+
153
+
163 rtp_error_t send_sdes_packet(const std::vector<uvgrtp::frame::rtcp_sdes_item>& items);
+
164
+
177 rtp_error_t send_app_packet(const char *name, uint8_t subtype, uint32_t payload_len, const uint8_t *payload);
+
178
+
193 rtp_error_t send_bye_packet(std::vector<uint32_t> ssrcs);
+
194
+
196 /* Return the latest RTCP packet received from participant of "ssrc"
+
197 * Return nullptr if we haven't received this kind of packet or if "ssrc" doesn't exist
+
198 *
+
199 * NOTE: Caller is responsible for deallocating the memory */
+
200 uvgrtp::frame::rtcp_sender_report *get_sender_packet(uint32_t ssrc);
+
201 uvgrtp::frame::rtcp_receiver_report *get_receiver_packet(uint32_t ssrc);
+
202 uvgrtp::frame::rtcp_sdes_packet *get_sdes_packet(uint32_t ssrc);
+
203 uvgrtp::frame::rtcp_app_packet *get_app_packet(uint32_t ssrc);
+
204
+
205 /* Somebody joined the multicast group the owner of this RTCP instance is part of
+
206 * Add it to RTCP participant list so we can start listening for reports
+
207 *
+
208 * "clock_rate" tells how much the RTP timestamp advances, this information is needed
+
209 * to calculate the interarrival jitter correctly. It has nothing do with our clock rate,
+
210 * (or whether we're even sending anything)
+
211 *
+
212 * Return RTP_OK on success and RTP_ERROR on error */
+
213 rtp_error_t add_initial_participant(uint32_t clock_rate);
+
214
+
215 /* Functions for updating various RTP sender statistics */
+
216 void sender_update_stats(const uvgrtp::frame::rtp_frame *frame);
+
217
+
218 /* If we've detected that our SSRC has collided with someone else's SSRC, we need to
+
219 * generate new random SSRC and reinitialize our own RTCP state.
+
220 * RTCP object still has the participants of "last session", we can use their SSRCs
+
221 * to detected new collision
+
222 *
+
223 * Return RTP_OK if reinitialization succeeded
+
224 * Return RTP_SSRC_COLLISION if our new SSRC has collided and we need to generate new SSRC */
+
225 rtp_error_t reset_rtcp_state(uint32_t ssrc);
+
226
+
227 /* Update various session statistics */
+
228 void update_session_statistics(const uvgrtp::frame::rtp_frame *frame);
+
229
+
230 /* Getter for interval_ms_, which is calculated by set_session_bandwidth */
+
231 uint32_t get_rtcp_interval_ms() const;
+
232
+
233 /* Set RTCP packet transmission interval in milliseconds
+
234 *
+
235 * Return RTP_OK if interval was set successfully
+
236 * Return RTP_INVALID_VALUE if new interval is invalid */
+
237 rtp_error_t set_rtcp_interval_ms(int32_t new_interval);
238
-
239 /* Store the following info in RTCP
-
240 * Local IP address
-
241 * Remote IP address
-
242 * Local port number for RTCP
-
243 * Destination port number for RTCP
-
244 * These are used when adding new participants and creating sockets for them */
+
239 /* Set total bandwidth for this session, called at the start
+
240 * If you want to set the interval manually later, use
+
241 * set_rtcp_interval_ms() function */
+
242 void set_session_bandwidth(uint32_t kbps);
+
243
+
244 std::shared_ptr<uvgrtp::socket> get_socket() const;
245
-
246 rtp_error_t set_network_addresses(std::string local_addr, std::string remote_addr,
-
247 uint16_t local_port, uint16_t dst_port);
-
248
-
249 /* Return SSRCs of all participants */
-
250 std::vector<uint32_t> get_participants() const;
+
246 /* Store the following info in RTCP
+
247 * Local IP address
+
248 * Remote IP address
+
249 * Local port number for RTCP
+
250 * Destination port number for RTCP
+
251 * These are used when adding new participants and creating sockets for them */
252
-
267 void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start);
-
268
-
269 /* Alternate way to get RTCP packets is to install a hook for them. So instead of
-
270 * polling an RTCP packet, user can install a function that is called when
-
271 * a specific RTCP packet is received. */
-
272
-
283 rtp_error_t install_sender_hook(void (*hook)(uvgrtp::frame::rtcp_sender_report *));
-
284
-
295 rtp_error_t install_sender_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_handler);
-
296
-
307 rtp_error_t install_receiver_hook(void (*hook)(uvgrtp::frame::rtcp_receiver_report *));
-
308
-
319 rtp_error_t install_receiver_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_handler);
-
320
-
331 rtp_error_t install_sdes_hook(void (*hook)(uvgrtp::frame::rtcp_sdes_packet *));
-
332
-
343 rtp_error_t install_sdes_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_handler);
-
344
-
355 rtp_error_t install_app_hook(void (*hook)(uvgrtp::frame::rtcp_app_packet *));
-
356
-
367 rtp_error_t install_app_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_app_packet>)> app_handler);
-
368
-
370 // These have been replaced by functions with unique_ptr in them
-
371 rtp_error_t install_sender_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_handler);
-
372 rtp_error_t install_receiver_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_handler);
-
373 rtp_error_t install_sdes_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_handler);
-
374 rtp_error_t install_app_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_app_packet>)> app_handler);
-
376
-
387 rtp_error_t install_send_app_hook(std::string app_name, std::function<std::unique_ptr<uint8_t[]>(uint8_t& subtype, uint32_t& payload_len)> app_sending_func);
-
388
-
396 rtp_error_t remove_all_hooks();
-
397
-
398 rtp_error_t remove_send_app_hook(std::string app_name);
-
399
-
401 /* Update RTCP-related sender statistics */
-
402 rtp_error_t update_sender_stats(size_t pkt_size);
-
403
-
404 /* Update RTCP-related receiver statistics */
-
405 static rtp_error_t recv_packet_handler(void *arg, int rce_flags, frame::rtp_frame **out);
+
253 rtp_error_t set_network_addresses(std::string local_addr, std::string remote_addr,
+
254 uint16_t local_port, uint16_t dst_port, bool ipv6);
+
255
+
256 /* Return SSRCs of all participants */
+
257 std::vector<uint32_t> get_participants() const;
+
259
+
274 void set_ts_info(uint64_t clock_start, uint32_t clock_rate, uint32_t rtp_ts_start);
+
275
+
276 /* Alternate way to get RTCP packets is to install a hook for them. So instead of
+
277 * polling an RTCP packet, user can install a function that is called when
+
278 * a specific RTCP packet is received. */
+
279
+
290 rtp_error_t install_sender_hook(void (*hook)(uvgrtp::frame::rtcp_sender_report *));
+
291
+
302 rtp_error_t install_sender_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_handler);
+
303
+
314 rtp_error_t install_receiver_hook(void (*hook)(uvgrtp::frame::rtcp_receiver_report *));
+
315
+
326 rtp_error_t install_receiver_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_handler);
+
327
+
338 rtp_error_t install_sdes_hook(void (*hook)(uvgrtp::frame::rtcp_sdes_packet *));
+
339
+
350 rtp_error_t install_sdes_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_handler);
+
351
+
362 rtp_error_t install_app_hook(void (*hook)(uvgrtp::frame::rtcp_app_packet *));
+
363
+
374 rtp_error_t install_app_hook(std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_app_packet>)> app_handler);
+
375
+
377 // These have been replaced by functions with unique_ptr in them
+
378 rtp_error_t install_sender_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_handler);
+
379 rtp_error_t install_receiver_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_handler);
+
380 rtp_error_t install_sdes_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_handler);
+
381 rtp_error_t install_app_hook(std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_app_packet>)> app_handler);
+
383
+
394 rtp_error_t install_send_app_hook(std::string app_name, std::function<std::unique_ptr<uint8_t[]>(uint8_t& subtype, uint32_t& payload_len)> app_sending_func);
+
395
+
403 rtp_error_t remove_all_hooks();
+
404
+
405 rtp_error_t remove_send_app_hook(std::string app_name);
406
-
407 /* Update RTCP-related sender statistics */
-
408 static rtp_error_t send_packet_handler_vec(void *arg, uvgrtp::buf_vec& buffers);
-
409
-
410 // the length field is the rtcp packet size measured in 32-bit words - 1
-
411 size_t rtcp_length_in_bytes(uint16_t length);
-
412
-
413 void set_payload_size(size_t mtu_size);
-
415
-
416 private:
-
417
-
418 rtp_error_t set_sdes_items(const std::vector<uvgrtp::frame::rtcp_sdes_item>& items);
+
408 /* Update RTCP-related sender statistics */
+
409 rtp_error_t update_sender_stats(size_t pkt_size);
+
410
+
411 /* Update RTCP-related receiver statistics */
+
412 static rtp_error_t recv_packet_handler(void *arg, int rce_flags, frame::rtp_frame **out);
+
413
+
414 /* Update RTCP-related sender statistics */
+
415 static rtp_error_t send_packet_handler_vec(void *arg, uvgrtp::buf_vec& buffers);
+
416
+
417 // the length field is the rtcp packet size measured in 32-bit words - 1
+
418 size_t rtcp_length_in_bytes(uint16_t length);
419
-
420 uint32_t size_of_ready_app_packets() const;
-
421 uint32_t size_of_apps_from_hook(std::vector< std::shared_ptr<rtcp_app_packet>> packets) const;
+
420 void set_payload_size(size_t mtu_size);
422
-
423 uint32_t size_of_compound_packet(uint16_t reports,
-
424 bool sr_packet, bool rr_packet, bool sdes_packet, uint32_t app_size, bool bye_packet) const;
-
425
-
426 /* read the header values from rtcp packet */
-
427 void read_rtcp_header(const uint8_t* buffer, size_t& read_ptr,
-
428 uvgrtp::frame::rtcp_header& header);
-
429 void read_reports(const uint8_t* buffer, size_t& read_ptr, size_t packet_end, uint8_t count,
-
430 std::vector<uvgrtp::frame::rtcp_report_block>& reports);
-
431
-
432 void read_ssrc(const uint8_t* buffer, size_t& read_ptr, uint32_t& out_ssrc);
-
433
-
434 /* Handle different kinds of incoming rtcp packets. The read header is passed to functions
-
435 which read rest of the frame type specific data.
-
436 * Return RTP_OK on success and RTP_ERROR on error */
-
437 rtp_error_t handle_sender_report_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
-
438 uvgrtp::frame::rtcp_header& header);
-
439 rtp_error_t handle_receiver_report_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
-
440 uvgrtp::frame::rtcp_header& header);
-
441 rtp_error_t handle_sdes_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
-
442 uvgrtp::frame::rtcp_header& header, uint32_t sender_ssrc);
-
443 rtp_error_t handle_bye_packet(uint8_t* buffer, size_t& read_ptr,
-
444 uvgrtp::frame::rtcp_header& header);
-
445 rtp_error_t handle_app_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
-
446 uvgrtp::frame::rtcp_header& header);
-
447
-
448 static void rtcp_runner(rtcp *rtcp);
-
449
-
450 static void rtcp_report_reader(rtcp *rtcp);
-
451
-
452 /* when we start the RTCP instance, we don't know what the SSRC of the remote is
-
453 * when an RTP packet is received, we must check if we've already received a packet
-
454 * from this sender and if not, create new entry to receiver_stats_ map */
-
455 bool is_participant(uint32_t ssrc) const;
+
423 private:
+
424
+
425 rtp_error_t set_sdes_items(const std::vector<uvgrtp::frame::rtcp_sdes_item>& items);
+
426
+
427 uint32_t size_of_ready_app_packets() const;
+
428 uint32_t size_of_apps_from_hook(std::vector< std::shared_ptr<rtcp_app_packet>> packets) const;
+
429
+
430 uint32_t size_of_compound_packet(uint16_t reports,
+
431 bool sr_packet, bool rr_packet, bool sdes_packet, uint32_t app_size, bool bye_packet) const;
+
432
+
433 /* read the header values from rtcp packet */
+
434 void read_rtcp_header(const uint8_t* buffer, size_t& read_ptr,
+
435 uvgrtp::frame::rtcp_header& header);
+
436 void read_reports(const uint8_t* buffer, size_t& read_ptr, size_t packet_end, uint8_t count,
+
437 std::vector<uvgrtp::frame::rtcp_report_block>& reports);
+
438
+
439 void read_ssrc(const uint8_t* buffer, size_t& read_ptr, uint32_t& out_ssrc);
+
440
+
441 /* Handle different kinds of incoming rtcp packets. The read header is passed to functions
+
442 which read rest of the frame type specific data.
+
443 * Return RTP_OK on success and RTP_ERROR on error */
+
444 rtp_error_t handle_sender_report_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
+
445 uvgrtp::frame::rtcp_header& header);
+
446 rtp_error_t handle_receiver_report_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
+
447 uvgrtp::frame::rtcp_header& header);
+
448 rtp_error_t handle_sdes_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
+
449 uvgrtp::frame::rtcp_header& header, uint32_t sender_ssrc);
+
450 rtp_error_t handle_bye_packet(uint8_t* buffer, size_t& read_ptr,
+
451 uvgrtp::frame::rtcp_header& header);
+
452 rtp_error_t handle_app_packet(uint8_t* buffer, size_t& read_ptr, size_t packet_end,
+
453 uvgrtp::frame::rtcp_header& header);
+
454
+
455 static void rtcp_runner(rtcp *rtcp);
456
-
457 //TODO: Resolve collision??
-
458 /* When we receive an RTP or RTCP packet, we need to check the source address and see if it's
-
459 * the same address where we've received packets before.
-
460 *
-
461 * If the address is new, it means we have detected an SSRC collision and the paket should
-
462 * be dropped We also need to check whether this SSRC matches with our own SSRC and if it does
-
463 * we need to send RTCP BYE and rejoin to the session */
-
464 bool collision_detected(uint32_t ssrc) const;
-
465
-
466 /* Move participant from initial_peers_ to participants_ */
-
467 rtp_error_t add_participant(uint32_t ssrc);
-
468
-
469 /* We've got a message from new source (the SSRC of the frame is not known to us)
-
470 * Initialize statistics for the peer and move it to participants_ */
-
471 rtp_error_t init_new_participant(const uvgrtp::frame::rtp_frame *frame);
+
457 static void rtcp_report_reader(rtcp *rtcp);
+
458
+
459 /* when we start the RTCP instance, we don't know what the SSRC of the remote is
+
460 * when an RTP packet is received, we must check if we've already received a packet
+
461 * from this sender and if not, create new entry to receiver_stats_ map */
+
462 bool is_participant(uint32_t ssrc) const;
+
463
+
464 //TODO: Resolve collision??
+
465 /* When we receive an RTP or RTCP packet, we need to check the source address and see if it's
+
466 * the same address where we've received packets before.
+
467 *
+
468 * If the address is new, it means we have detected an SSRC collision and the paket should
+
469 * be dropped We also need to check whether this SSRC matches with our own SSRC and if it does
+
470 * we need to send RTCP BYE and rejoin to the session */
+
471 bool collision_detected(uint32_t ssrc) const;
472
-
473 /* Initialize the RTP Sequence related stuff of peer
-
474 * This function assumes that the peer already exists in the participants_ map */
-
475 rtp_error_t init_participant_seq(uint32_t ssrc, uint16_t base_seq);
-
476
-
477 /* Update the SSRC's sequence related data in participants_ map
-
478 *
-
479 * Return RTP_OK if the received packet was OK
-
480 * Return RTP_GENERIC_ERROR if it wasn't and
-
481 * packet-related statistics should not be updated */
-
482 rtp_error_t update_participant_seq(uint32_t ssrc, uint16_t seq);
+
473 /* Move participant from initial_peers_ to participants_ */
+
474 rtp_error_t add_participant(uint32_t ssrc);
+
475
+
476 /* We've got a message from new source (the SSRC of the frame is not known to us)
+
477 * Initialize statistics for the peer and move it to participants_ */
+
478 rtp_error_t init_new_participant(const uvgrtp::frame::rtp_frame *frame);
+
479
+
480 /* Initialize the RTP Sequence related stuff of peer
+
481 * This function assumes that the peer already exists in the participants_ map */
+
482 rtp_error_t init_participant_seq(uint32_t ssrc, uint16_t base_seq);
483
-
484 /* Update the RTCP bandwidth variables
+
484 /* Update the SSRC's sequence related data in participants_ map
485 *
-
486 * "pkt_size" tells how much rtcp_byte_count_
-
487 * should be increased before calculating the new average */
-
488 void update_rtcp_bandwidth(size_t pkt_size);
-
489
-
490 /* Update average RTCP packet size variable
-
491 * packet_size is the size of received RTCP packet in octets */
-
492 void update_avg_rtcp_size(uint64_t packet_size);
-
493
-
494 /* Calculate the RTCP report interval in seconds
-
495 * rtcp_bw is given in kbps
-
496 * Defined in RFC3550 Appendix A.7 */
-
497 double rtcp_interval(int members, int senders,
-
498 double rtcp_bw, bool we_sent, double avg_rtcp_size, bool red_min, bool randomisation);
-
499
-
500 /* RTCP runner keeps track of ssrcs and how long they have been silent.
-
501 * By default a source get timed out if it has been silent for 25 seconds
-
502 * If an ssrc is timed out, this function removes it from participants_ map and
-
503 * updates any other infos */
-
504 rtp_error_t remove_timeout_ssrc(uint32_t ssrc);
-
505
-
506 /* Because struct statistics contains uvgRTP clock object we cannot
-
507 * zero it out without compiler complaining about it so all the fields
-
508 * must be set to zero manually */
-
509 void zero_stats(uvgrtp::sender_statistics *stats);
-
510
-
511 void zero_stats(uvgrtp::receiver_statistics *stats);
+
486 * Return RTP_OK if the received packet was OK
+
487 * Return RTP_GENERIC_ERROR if it wasn't and
+
488 * packet-related statistics should not be updated */
+
489 rtp_error_t update_participant_seq(uint32_t ssrc, uint16_t seq);
+
490
+
491 /* Update the RTCP bandwidth variables
+
492 *
+
493 * "pkt_size" tells how much rtcp_byte_count_
+
494 * should be increased before calculating the new average */
+
495 void update_rtcp_bandwidth(size_t pkt_size);
+
496
+
497 /* Update average RTCP packet size variable
+
498 * packet_size is the size of received RTCP packet in octets */
+
499 void update_avg_rtcp_size(uint64_t packet_size);
+
500
+
501 /* Calculate the RTCP report interval in seconds
+
502 * rtcp_bw is given in kbps
+
503 * Defined in RFC3550 Appendix A.7 */
+
504 double rtcp_interval(int members, int senders,
+
505 double rtcp_bw, bool we_sent, double avg_rtcp_size, bool red_min, bool randomisation);
+
506
+
507 /* RTCP runner keeps track of ssrcs and how long they have been silent.
+
508 * By default a source get timed out if it has been silent for 25 seconds
+
509 * If an ssrc is timed out, this function removes it from participants_ map and
+
510 * updates any other infos */
+
511 rtp_error_t remove_timeout_ssrc(uint32_t ssrc);
512
-
513 /* Takes ownership of the frame */
-
514 rtp_error_t send_rtcp_packet_to_participants(uint8_t* frame, uint32_t frame_size, bool encrypt);
-
515
-
516 void free_participant(std::unique_ptr<rtcp_participant> participant);
+
513 /* Because struct statistics contains uvgRTP clock object we cannot
+
514 * zero it out without compiler complaining about it so all the fields
+
515 * must be set to zero manually */
+
516 void zero_stats(uvgrtp::sender_statistics *stats);
517
-
518 void cleanup_participants();
+
518 void zero_stats(uvgrtp::receiver_statistics *stats);
519
-
520 /* Secure RTCP context */
-
521 std::shared_ptr<uvgrtp::srtcp> srtcp_;
+
520 /* Takes ownership of the frame */
+
521 rtp_error_t send_rtcp_packet_to_participants(uint8_t* frame, uint32_t frame_size, bool encrypt);
522
-
523 /* RTP context flags */
-
524 int rce_flags_;
-
525
-
526 /* are we a sender (and possible a receiver) or just a receiver */
-
527 int our_role_;
-
528
-
529 /* TODO: time_t?? */
-
530 // TODO: Check these, they don't seem to be used
-
531 size_t tp_; /* the last time an RTCP packet was transmitted */
-
532 size_t tc_; /* the current time */
-
533 size_t tn_; /* the next scheduled transmission time of an RTCP packet */
-
534 size_t pmembers_; /* the estimated number of session members at the time tn was last recomputed */
-
535 size_t members_; /* the most current estimate for the number of session members */
-
536 size_t senders_; /* the most current estimate for the number of senders in the session */
-
537
-
538 /* Total session bandwidth. RTCP bandwidth will be set to 5 % of this */
-
539 uint32_t total_bandwidth_;
-
540
-
541 /* The target RTCP bandwidth, i.e., the total bandwidth
-
542 * that will be used for RTCP packets by all members of this session,
-
543 * in octets per second. This will be a specified fraction of the
-
544 * "session bandwidth" parameter supplied to the application at startup. */
-
545 double rtcp_bandwidth_;
-
546
-
547 /* "Minimum" value for RTCP transmission interval, depends on the session bandwidth
-
548 * Actual interval can be 50 % smaller due to randomisation */
-
549 uint32_t reduced_minimum_;
-
550
-
551 /* Flag that is true if the application has sent data since
-
552 * the 2nd previous RTCP report was transmitted. */
-
553 // TODO: Only set, never read
-
554 bool we_sent_;
-
555
-
556 /* Store sender and receiver info, this is needed when calling
-
557 * add_participant dynamically (i.e. after initializing the stream) */
-
558 std::string local_addr_;
-
559 std::string remote_addr_;
-
560 uint16_t local_port_;
-
561 uint16_t dst_port_;
+
523 void free_participant(std::unique_ptr<rtcp_participant> participant);
+
524
+
525 void cleanup_participants();
+
526
+
527 /* Secure RTCP context */
+
528 std::shared_ptr<uvgrtp::srtcp> srtcp_;
+
529
+
530 /* RTP context flags */
+
531 int rce_flags_;
+
532
+
533 /* are we a sender (and possible a receiver) or just a receiver */
+
534 int our_role_;
+
535
+
536 /* TODO: time_t?? */
+
537 // TODO: Check these, they don't seem to be used
+
538 size_t tp_; /* the last time an RTCP packet was transmitted */
+
539 size_t tc_; /* the current time */
+
540 size_t tn_; /* the next scheduled transmission time of an RTCP packet */
+
541 size_t pmembers_; /* the estimated number of session members at the time tn was last recomputed */
+
542 size_t members_; /* the most current estimate for the number of session members */
+
543 size_t senders_; /* the most current estimate for the number of senders in the session */
+
544
+
545 /* Total session bandwidth. RTCP bandwidth will be set to 5 % of this */
+
546 uint32_t total_bandwidth_;
+
547
+
548 /* The target RTCP bandwidth, i.e., the total bandwidth
+
549 * that will be used for RTCP packets by all members of this session,
+
550 * in octets per second. This will be a specified fraction of the
+
551 * "session bandwidth" parameter supplied to the application at startup. */
+
552 double rtcp_bandwidth_;
+
553
+
554 /* "Minimum" value for RTCP transmission interval, depends on the session bandwidth
+
555 * Actual interval can be 50 % smaller due to randomisation */
+
556 uint32_t reduced_minimum_;
+
557
+
558 /* Flag that is true if the application has sent data since
+
559 * the 2nd previous RTCP report was transmitted. */
+
560 // TODO: Only set, never read
+
561 bool we_sent_;
562
-
563 /* The average compound RTCP packet size, in octets,
-
564 * over all RTCP packets sent and received by this participant. The
-
565 * size includes lower-layer transport and network protocol headers
-
566 * (e.g., UDP and IP) as explained in Section 6.2 */
-
567 // TODO: Only set, never read
-
568 size_t avg_rtcp_pkt_pize_;
+
563 /* Store sender and receiver info, this is needed when calling
+
564 * add_participant dynamically (i.e. after initializing the stream) */
+
565 std::string local_addr_;
+
566 std::string remote_addr_;
+
567 uint16_t local_port_;
+
568 uint16_t dst_port_;
569
-
570 /* Average RTCP packet size in octets.
-
571 * Initialized to 64 */
-
572 uint64_t avg_rtcp_size_;
-
573
-
574 /* Number of RTCP packets and bytes sent and received by this participant */
-
575 // TODO: Only set, never read
-
576 size_t rtcp_pkt_count_;
-
577 size_t rtcp_byte_count_;
-
578
-
579 /* Number of RTCP packets sent */
-
580 uint32_t rtcp_pkt_sent_count_;
-
581
-
582 /* Flag that is true if the application has not yet sent an RTCP packet. */
-
583 // TODO: Only set, never read
-
584 bool initial_;
+
570 /* The average compound RTCP packet size, in octets,
+
571 * over all RTCP packets sent and received by this participant. The
+
572 * size includes lower-layer transport and network protocol headers
+
573 * (e.g., UDP and IP) as explained in Section 6.2 */
+
574 // TODO: Only set, never read
+
575 size_t avg_rtcp_pkt_pize_;
+
576
+
577 /* Average RTCP packet size in octets.
+
578 * Initialized to 64 */
+
579 uint64_t avg_rtcp_size_;
+
580
+
581 /* Number of RTCP packets and bytes sent and received by this participant */
+
582 // TODO: Only set, never read
+
583 size_t rtcp_pkt_count_;
+
584 size_t rtcp_byte_count_;
585
-
586 /* Copy of our own current SSRC */
-
587 std::shared_ptr<std::atomic_uint> ssrc_;
+
586 /* Number of RTCP packets sent */
+
587 uint32_t rtcp_pkt_sent_count_;
588
-
589 /* NTP timestamp associated with initial RTP timestamp (aka t = 0) */
-
590 uint64_t clock_start_;
-
591
-
592 /* Clock rate of the media ie. how fast does the time increase */
-
593 uint32_t clock_rate_;
-
594
-
595 /* The first value of RTP timestamp (aka t = 0) */
-
596 uint32_t rtp_ts_start_;
-
597
-
598 std::map<uint32_t, std::unique_ptr<rtcp_participant>> participants_;
-
599 uint8_t num_receivers_; // maximum is 32 at the moment (5 bits)
-
600
-
601 /* Address of the socket that we are sending data to */
-
602 sockaddr_in socket_address_ = {};
-
603
+
589 /* Flag that is true if the application has not yet sent an RTCP packet. */
+
590 // TODO: Only set, never read
+
591 bool initial_;
+
592
+
593 /* Copy of our own current SSRC */
+
594 std::shared_ptr<std::atomic_uint> ssrc_;
+
595
+
596 /* NTP timestamp associated with initial RTP timestamp (aka t = 0) */
+
597 uint64_t clock_start_;
+
598
+
599 /* Clock rate of the media ie. how fast does the time increase */
+
600 uint32_t clock_rate_;
+
601
+
602 /* The first value of RTP timestamp (aka t = 0) */
+
603 uint32_t rtp_ts_start_;
604
-
605 /* Map for keeping track of sources for timeouts
-
606 * First number is the sources ssrc
-
607 * Second number is how many milliseconds it has been silent*/
-
608 std::map<uint32_t, uint32_t> ms_since_last_rep_;
-
609
-
610 /* statistics for RTCP Sender and Receiver Reports */
-
611 struct sender_statistics our_stats;
+
605 std::map<uint32_t, std::unique_ptr<rtcp_participant>> participants_;
+
606 uint8_t num_receivers_; // maximum is 32 at the moment (5 bits)
+
607 bool ipv6_;
+
608
+
609 /* Address of the socket that we are sending data to */
+
610 sockaddr_in socket_address_;
+
611 sockaddr_in6 socket_address_ipv6_;
612
-
613 /* If we expect frames from remote but haven't received anything from remote yet,
-
614 * the participant resides in this vector until he's moved to participants_ */
-
615 std::vector<std::unique_ptr<rtcp_participant>> initial_participants_;
-
616
-
617
+
613
+
614 /* Map for keeping track of sources for timeouts
+
615 * First number is the sources ssrc
+
616 * Second number is how many milliseconds it has been silent*/
+
617 std::map<uint32_t, uint32_t> ms_since_last_rep_;
618
-
619 void (*sender_hook_)(uvgrtp::frame::rtcp_sender_report *);
-
620 void (*receiver_hook_)(uvgrtp::frame::rtcp_receiver_report *);
-
621 void (*sdes_hook_)(uvgrtp::frame::rtcp_sdes_packet *);
-
622 void (*app_hook_)(uvgrtp::frame::rtcp_app_packet *);
-
623
-
624 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_hook_f_;
-
625 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_hook_u_;
-
626 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_hook_f_;
-
627 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_hook_u_;
-
628 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_hook_f_;
-
629 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_hook_u_;
-
630 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_app_packet>)> app_hook_f_;
-
631 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_app_packet>)> app_hook_u_;
+
619 /* statistics for RTCP Sender and Receiver Reports */
+
620 struct sender_statistics our_stats;
+
621
+
622 /* If we expect frames from remote but haven't received anything from remote yet,
+
623 * the participant resides in this vector until he's moved to participants_ */
+
624 std::vector<std::unique_ptr<rtcp_participant>> initial_participants_;
+
625
+
626
+
627
+
628 void (*sender_hook_)(uvgrtp::frame::rtcp_sender_report *);
+
629 void (*receiver_hook_)(uvgrtp::frame::rtcp_receiver_report *);
+
630 void (*sdes_hook_)(uvgrtp::frame::rtcp_sdes_packet *);
+
631 void (*app_hook_)(uvgrtp::frame::rtcp_app_packet *);
632
-
633 std::mutex sr_mutex_;
-
634 std::mutex rr_mutex_;
-
635 std::mutex sdes_mutex_;
-
636 std::mutex app_mutex_;
-
637 mutable std::mutex participants_mutex_;
-
638 std::mutex send_app_mutex_;
-
639
-
640 std::unique_ptr<std::thread> report_generator_;
-
641 std::unique_ptr<std::thread> report_reader_;
-
642 std::shared_ptr<uvgrtp::socket> rtcp_socket_;
-
643
-
644 bool is_active() const
-
645 {
-
646 return active_;
-
647 }
+
633 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_hook_f_;
+
634 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sender_report>)> sr_hook_u_;
+
635 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_hook_f_;
+
636 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_receiver_report>)> rr_hook_u_;
+
637 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_hook_f_;
+
638 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_sdes_packet>)> sdes_hook_u_;
+
639 std::function<void(std::shared_ptr<uvgrtp::frame::rtcp_app_packet>)> app_hook_f_;
+
640 std::function<void(std::unique_ptr<uvgrtp::frame::rtcp_app_packet>)> app_hook_u_;
+
641
+
642 std::mutex sr_mutex_;
+
643 std::mutex rr_mutex_;
+
644 std::mutex sdes_mutex_;
+
645 std::mutex app_mutex_;
+
646 mutable std::mutex participants_mutex_;
+
647 std::mutex send_app_mutex_;
648
-
649 bool active_;
-
650
-
651 std::atomic<uint32_t> interval_ms_;
+
649 std::unique_ptr<std::thread> report_generator_;
+
650 std::unique_ptr<std::thread> report_reader_;
+
651 std::shared_ptr<uvgrtp::socket> rtcp_socket_;
652
-
653 std::shared_ptr<uvgrtp::rtp> rtp_ptr_;
-
654
-
655 std::mutex packet_mutex_;
-
656
-
657 // messages waiting to be sent
-
658 std::vector<uvgrtp::frame::rtcp_sdes_item> ourItems_; // always sent
-
659 std::vector<uint32_t> bye_ssrcs_; // sent once
-
660
-
661 std::map<std::string, std::deque<rtcp_app_packet>> app_packets_; // sent one at a time per name
-
662 // APPs for hook
-
663 std::multimap<std::string, std::function <std::unique_ptr<uint8_t[]>(uint8_t& subtype, uint32_t& payload_len)>> outgoing_app_hooks_;
-
664
-
665 bool hooked_app_;
-
666
-
667
-
668 uvgrtp::frame::rtcp_sdes_item cnameItem_;
-
669 char cname_[255];
-
670
-
671 size_t mtu_size_;
-
672 };
-
673}
-
674
-
675namespace uvg_rtp = uvgrtp;
-
uvgrtp::rtcp
RTCP instance handles all incoming and outgoing RTCP traffic, including report generation.
Definition: rtcp.hh:111
+
653 bool is_active() const
+
654 {
+
655 return active_;
+
656 }
+
657
+
658 bool active_;
+
659
+
660 std::atomic<uint32_t> interval_ms_;
+
661
+
662 std::shared_ptr<uvgrtp::rtp> rtp_ptr_;
+
663
+
664 std::mutex packet_mutex_;
+
665
+
666 // messages waiting to be sent
+
667 std::vector<uvgrtp::frame::rtcp_sdes_item> ourItems_; // always sent
+
668 std::vector<uint32_t> bye_ssrcs_; // sent once
+
669
+
670 std::map<std::string, std::deque<rtcp_app_packet>> app_packets_; // sent one at a time per name
+
671 // APPs for hook
+
672 std::multimap<std::string, std::function <std::unique_ptr<uint8_t[]>(uint8_t& subtype, uint32_t& payload_len)>> outgoing_app_hooks_;
+
673
+
674 bool hooked_app_;
+
675
+
676
+
677 uvgrtp::frame::rtcp_sdes_item cnameItem_;
+
678 char cname_[255];
+
679
+
680 size_t mtu_size_;
+
681 };
+
682}
+
683
+
684namespace uvg_rtp = uvgrtp;
+
uvgrtp::rtcp
RTCP instance handles all incoming and outgoing RTCP traffic, including report generation.
Definition: rtcp.hh:118
uvgrtp::rtcp::install_receiver_hook
rtp_error_t install_receiver_hook(void(*hook)(uvgrtp::frame::rtcp_receiver_report *))
Install an RTCP Receiver Report hook.
uvgrtp::rtcp::send_app_packet
rtp_error_t send_app_packet(const char *name, uint8_t subtype, uint32_t payload_len, const uint8_t *payload)
Send an RTCP APP packet.
uvgrtp::rtcp::install_receiver_hook
rtp_error_t install_receiver_hook(std::function< void(std::unique_ptr< uvgrtp::frame::rtcp_receiver_report >)> rr_handler)
Install an RTCP Receiver Report hook.
diff --git a/docs/html/session_8hh_source.html b/docs/html/session_8hh_source.html index 4cef602f..4f40137b 100644 --- a/docs/html/session_8hh_source.html +++ b/docs/html/session_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -130,7 +130,7 @@
127}
128
129namespace uvg_rtp = uvgrtp;
-
uvgrtp::media_stream
The media_stream is an entity which represents one RTP stream.
Definition: media_stream.hh:49
+
uvgrtp::media_stream
The media_stream is an entity which represents one RTP stream.
Definition: media_stream.hh:50
uvgrtp::session
Provides ZRTP synchronization and can be used to create uvgrtp::media_stream objects.
Definition: session.hh:26
uvgrtp::session::create_stream
uvgrtp::media_stream * create_stream(uint16_t port, rtp_format_t fmt, int rce_flags)
Create a unidirectional media_stream for an RTP session.
uvgrtp::session::create_stream
uvgrtp::media_stream * create_stream(uint16_t src_port, uint16_t dst_port, rtp_format_t fmt, int rce_flags)
Create a uni- or bidirectional media stream.
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet-members.html index da48d0bc..d7dea54b 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet.html index f4c923e7..1c707310 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__app__packet.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__header-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__header-members.html index 2bd5ca6b..47d29f81 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__header-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__header-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__header.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__header.html index f9f33b0c..691645ea 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__header.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__header.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report-members.html index a94cd0b6..7c898e67 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report.html index aea229cd..a1fdbeb9 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__receiver__report.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block-members.html index 9121d027..9fc2b7ec 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block.html index e1aad227..87200a21 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__report__block.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk-members.html index ec12a804..f17de046 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk.html index f012df61..e70906d3 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__chunk.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item-members.html index 044309f1..1f3f5dda 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item.html index e72a2799..db52264a 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__item.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet-members.html index e3781b2e..74c75c35 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet.html index efe53f3b..9c39e1ee 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sdes__packet.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info-members.html index d5150242..9e25b6c4 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info.html index 69f94afb..ac87a0be 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__info.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report-members.html index c2a1149e..108bc350 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report.html b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report.html index 1817e1ab..b75e8175 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtcp__sender__report.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtp__frame-members.html b/docs/html/structuvgrtp_1_1frame_1_1rtp__frame-members.html index e5719e0e..74cae8a2 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtp__frame-members.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtp__frame-members.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/structuvgrtp_1_1frame_1_1rtp__frame.html b/docs/html/structuvgrtp_1_1frame_1_1rtp__frame.html index a55c9f1d..c5e68ff6 100644 --- a/docs/html/structuvgrtp_1_1frame_1_1rtp__frame.html +++ b/docs/html/structuvgrtp_1_1frame_1_1rtp__frame.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/util_8hh.html b/docs/html/util_8hh.html index 936efb49..9ab8d6f8 100644 --- a/docs/html/util_8hh.html +++ b/docs/html/util_8hh.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
@@ -318,7 +318,7 @@

RCC_CLOCK_RATE 

Change uvgRTP's clock rate in RTP header and RTP timestamp calculations

RCC_MTU_SIZE 

Set a maximum value for the Ethernet frame size assumed by uvgRTP.

-

Default is 1492, from this IPv4 and UDP, and RTP headers are removed, giving a payload size of 1452 bytes.

+

Default is 1492, from this IP and UDP, and RTP headers are removed, giving a payload size of 1452 bytes.

If application wishes to use small UDP datagram, it can set MTU size to, for example, 500 bytes or if it wishes to use jumbo frames, it can set the MTU size to 9000 bytes

RCC_FPS_NUMERATOR 

Set the numerator of frame rate used by uvgRTP.

diff --git a/docs/html/util_8hh_source.html b/docs/html/util_8hh_source.html index bd230d6f..9b4eb5c3 100644 --- a/docs/html/util_8hh_source.html +++ b/docs/html/util_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/version_8hh_source.html b/docs/html/version_8hh_source.html index aaa20294..558eaa85 100644 --- a/docs/html/version_8hh_source.html +++ b/docs/html/version_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/docs/html/wrapper__c_8hh_source.html b/docs/html/wrapper__c_8hh_source.html index 878885bc..ff6449c5 100644 --- a/docs/html/wrapper__c_8hh_source.html +++ b/docs/html/wrapper__c_8hh_source.html @@ -21,7 +21,7 @@ -
uvgRTP 2.2.0 +
uvgRTP 2.3.0
An open-source library for RTP/SRTP media delivery
diff --git a/src/media_stream.cc b/src/media_stream.cc index 216e4aad..77d589e7 100644 --- a/src/media_stream.cc +++ b/src/media_stream.cc @@ -449,13 +449,14 @@ rtp_error_t uvgrtp::media_stream::start_components() } } - if (rce_flags_ & RCE_SRTP_AUTHENTICATE_RTP) + if (rce_flags_ & RCE_SRTP_AUTHENTICATE_RTP) { if (ipv6_) { rtp_->set_payload_size(MAX_IPV6_MEDIA_PAYLOAD - UVG_AUTH_TAG_LENGTH); } else { rtp_->set_payload_size(MAX_IPV4_MEDIA_PAYLOAD - UVG_AUTH_TAG_LENGTH); } + } initialized_ = true; return reception_flow_->start(socket_, rce_flags_);