This is the complete list of members for uvgrtp::context, including all inherited members.
| context() | uvgrtp::context | |
| create_session(std::string address) | uvgrtp::context | |
| create_session(std::string remote_addr, std::string local_addr) | uvgrtp::context | |
| create_session(std::pair< std::string, std::string > addresses) | uvgrtp::context | |
| create_session(std::string address) | uvgrtp::context | |
| crypto_enabled() const | uvgrtp::context | |
| destroy_session(uvgrtp::session *session) | uvgrtp::context | |
| ~context() | uvgrtp::context | |
|
- uvgRTP 2.3.0
+ uvgRTP 3.0.0
An open-source library for RTP/SRTP media delivery
|
@@ -94,12 +94,12 @@
||
| ~context () | ||
| RTP context destructor. | ||
| uvgrtp::session * | create_session (std::pair< std::string, std::string > addresses) | |
| Create a new RTP session between two IP addresses. | ||
| uvgrtp::session * | create_session (std::string address) | |
| Create a new RTP session. | ||
| uvgrtp::session * | create_session (std::string remote_addr, std::string local_addr) | |
| Create a new RTP session. | ||
| rtp_error_t | destroy_session (uvgrtp::session *session) | |
| Destroy RTP session and all of its media streams. | ||
| ( | -std::string | -address | ) | +std::pair< std::string, std::string > | +addresses | ) |
Create a new RTP session.
+Create a new RTP session between two IP addresses.
| address | IP address of the remote participant |
| addresses | Local and remote IP address for session as a pair |
| uvgrtp::session | On success |
| nullptr | If "address" is empty or memory allocation failed |
| nullptr | If memory allocation failed |
Create a new RTP session.
-If UDP holepunching should be utilized, in addition to remote IP address, the caller must also provide local IP address where uvgRTP should bind itself to. If you are using uvgRTP for unidirectional streaming, please take a look at RCE_HOLEPUNCH_KEEPALIVE
| remote_addr | IP address of the remote participant |
| local_addr | IP address of a local interface |
| address | IP address of the remote participant |
| uvgrtp::session | On success |
| nullptr | If memory allocation failed |
| nullptr | If "address" is empty or memory allocation failed |
| add_srtp_ctx(uint8_t *key, uint8_t *salt) | uvgrtp::media_stream | |
| configure_ctx(int rcc_flag, ssize_t value) | uvgrtp::media_stream | |
| get_ipv6() const (defined in uvgrtp::media_stream) | uvgrtp::media_stream | |
| get_configuration_value(int rcc_flag) | uvgrtp::media_stream | |
| get_rtcp() | uvgrtp::media_stream | |
| get_ssrc() const | uvgrtp::media_stream | |
| install_receive_hook(void *arg, void(*hook)(void *, uvgrtp::frame::rtp_frame *)) | 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 | |
| start_zrtp() | uvgrtp::media_stream |
Public Member Functions | |
| rtp_error_t | start_zrtp () |
| Start the ZRTP negotiation manually. | |
| rtp_error_t | add_srtp_ctx (uint8_t *key, uint8_t *salt) |
| Add keying information for user-managed SRTP session. | |
| rtp_error_t | configure_ctx (int rcc_flag, ssize_t value) |
| Configure the media stream, see RTP_CTX_CONFIGURATION_FLAGS for more details. | |
| int | get_configuration_value (int rcc_flag) |
| Get the values associated with configuration flags, see RTP_CTX_CONFIGURATION_FLAGS for more details. | |
| uvgrtp::rtcp * | get_rtcp () |
| Get pointer to the RTCP object of the media stream. | |
| 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 |
The media_stream is an entity which represents one RTP stream.
@@ -137,7 +140,7 @@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 50 of file media_stream.hh.
+Definition at line 53 of file media_stream.hh.
| int uvgrtp::media_stream::get_configuration_value | +( | +int | +rcc_flag | ) | ++ |
Get the values associated with configuration flags, see RTP_CTX_CONFIGURATION_FLAGS for more details.
+| int | value on success |
| -1 | on error |
| rtp_error_t uvgrtp::media_stream::start_zrtp | +( | +) | ++ |
Start the ZRTP negotiation manually.
+There is two ways to use ZRTP.
| RTP_OK | On success |
| RTP_TIMEOUT | if ZRTP timed out |
| RTP_GENERIC_ERROR | on other errors |
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 118 of file rtcp.hh.
+Definition at line 120 of file rtcp.hh.
| rtp_error_t uvgrtp::rtcp::remove_send_app_hook | +( | +std::string | +app_name | ) | ++ |
Remove a hook for sending APP packets *.
+| app_name | name of the APP packet hook. Max 4 chars |
| RTP_OK | on success |
| RTP_INVALID_VALUE | if hook with given app_name was not found |
Definition at line 291 of file util.hh.
+Definition at line 281 of file util.hh.
@@ -410,9 +424,11 @@Paces the sending of frame fragments within frame interval (default 1/30 s)
Use a single UDP port for both RTP and RTCP transmission (default RTCP port is +1)
+Definition at line 164 of file util.hh.
+Definition at line 165 of file util.hh.
@@ -489,7 +505,7 @@Definition at line 137 of file util.hh.
@@ -563,6 +579,8 @@H.266/VVC.
V3C.
+Definition at line 62 of file util.hh.
diff --git a/docs/html/util_8hh_source.html b/docs/html/util_8hh_source.html index 9b4eb5c3..f9af680f 100644 --- a/docs/html/util_8hh_source.html +++ b/docs/html/util_8hh_source.html @@ -21,7 +21,7 @@