Add Haivision SRT protocol

The protocol requires libsrt (https://github.com/Haivision/srt) to be
installed

Signed-off-by: Sven Dueking <sven.dueking@nablet.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

6 files changed, 696 insertions, 0 deletions

diff --git a/Changelog b/Changelog
index 0d20cd47df..53562a1aa0 100644
--- a/Changelog
+++ b/Changelog
@@ -21,6 +21,7 @@ version <next>:
 - NVIDIA CUVID-accelerated H.264 and HEVC decoding
 - Intel QSV-accelerated overlay filter
 - AV1 Support through libaom
+- Haivision SRT protocol via libsrt
 
 
 version 12:
diff --git a/configure b/configure
index 90fb6f07ca..7612a6052c 100755
--- a/configure
+++ b/configure
@@ -213,6 +213,7 @@ External library support:
   --enable-libschroedinger   Dirac video encoding/decoding
   --enable-libsnappy         snappy compression
   --enable-libspeex          Speex audio encoding/decoding
+  --enable-libsrt            Haivision SRT protocol
   --enable-libtheora         Theora video encoding/decoding
   --enable-libtwolame        MP2 audio encoding
   --enable-libvo-aacenc      AAC audio encoding
@@ -1374,6 +1375,7 @@ EXTERNAL_LIBRARY_LIST="
     libschroedinger
     libsnappy
     libspeex
+    libsrt
     libtheora
     libtwolame
     libvorbis
@@ -2525,6 +2527,8 @@ librtmpt_protocol_deps="librtmp"
 librtmpte_protocol_deps="librtmp"
 mmsh_protocol_select="http_protocol"
 mmst_protocol_select="network"
+libsrt_protocol_deps="libsrt"
+libsrt_protocol_select="network"
 rtmp_protocol_conflict="librtmp_protocol"
 rtmp_protocol_select="tcp_protocol"
 rtmp_protocol_suggest="zlib"
@@ -4674,6 +4678,7 @@ enabled librtmp           && require_pkg_config librtmp librtmp librtmp/rtmp.h R
 enabled libschroedinger   && require_pkg_config libschroedinger schroedinger-1.0 schroedinger/schro.h schro_init
 enabled libsnappy         && require libsnappy snappy-c.h snappy_compress -lsnappy
 enabled libspeex          && require_pkg_config libspeex speex speex/speex.h speex_decoder_init
+enabled libsrt            && require_pkg_config libsrt "srt >= 1.2.0" srt/srt.h srt_socket
 enabled libtheora         && require libtheora theora/theoraenc.h th_info_init -ltheoraenc -ltheoradec -logg
 enabled libtwolame        && require libtwolame twolame.h twolame_init -ltwolame
 enabled libvo_aacenc      && require libvo_aacenc vo-aacenc/voAAC.h voGetAACEncAPI -lvo-aacenc
diff --git a/doc/protocols.texi b/doc/protocols.texi
index c136c74e41..e2d06a0675 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -655,6 +655,146 @@ To play back the first stream announced on one the default IPv6 SAP multicast ad
 avplay sap://[ff0e::2:7ffe]
 @end example
 
+@section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The supported syntax for a SRT URL is:
+@example
+srt://@var{hostname}:@var{port}[?@var{options}]
+@end example
+
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.
+
+or
+
+@example
+@var{options} srt://@var{hostname}:@var{port}
+@end example
+
+@var{options} contains a list of '-@var{key} @var{val}'
+options.
+
+This protocol accepts the following options.
+
+@table @option
+@item connect_timeout
+Connection timeout; SRT cannot connect for RTT > 1500 msec
+(2 handshake exchanges) with the default connect timeout of
+3 seconds. This option applies to the caller and rendezvous
+connection modes. The connect timeout is 10 times the value
+set for the rendezvous mode (which can be used as a
+workaround for this connection problem with earlier versions).
+
+@item ffs=@var{bytes}
+Flight Flag Size (Window Size), in bytes. FFS is actually an
+internal parameter and you should set it to not less than
+@option{recv_buffer_size} and @option{mss}. The default value
+is relatively large, therefore unless you set a very large receiver buffer,
+you do not need to change this option. Default value is 25600.
+
+@item inputbw=@var{bytes/seconds}
+Sender nominal input rate, in bytes per seconds. Used along with
+@option{oheadbw}, when @option{maxbw} is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent
+along with the main media stream:
+@option{inputbw} * (100 + @option{oheadbw}) / 100
+if @option{inputbw} is not set while @option{maxbw} is set to
+relative (0), the actual input rate is evaluated inside
+the library. Default value is 0.
+
+@item iptos=@var{tos}
+IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+@item ipttl=@var{ttl}
+IP Time To Live. Applies to sender only. Default value is 64.
+
+@item listen_timeout
+Set socket listen timeout.
+
+@item maxbw=@var{bytes/seconds}
+Maximum sending bandwidth, in bytes per seconds.
+-1 infinite (CSRTCC limit is 30mbps)
+0 relative to input rate (see @option{inputbw})
+>0 absolute limit value
+Default value is 0 (relative)
+
+@item mode=@var{caller|listener|rendezvous}
+Connection mode.
+@option{caller} opens client connection.
+@option{listener} starts server to listen for incoming connections.
+@option{rendezvous} use Rendez-Vous connection mode.
+Default value is caller.
+
+@item mss=@var{bytes}
+Maximum Segment Size, in bytes. Used for buffer allocation
+and rate calculation using a packet counter assuming fully
+filled packets. The smallest MSS between the peers is
+used. This is 1500 by default in the overall internet.
+This is the maximum size of the UDP packet and can be
+only decreased, unless you have some unusual dedicated
+network settings. Default value is 1500.
+
+@item nakreport=@var{1|0}
+If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
+periodically until a lost packet is retransmitted or
+intentionally dropped. Default value is 1.
+
+@item oheadbw=@var{percents}
+Recovery bandwidth overhead above input rate, in percents.
+See @option{inputbw}. Default value is 25%.
+
+@item passphrase=@var{string}
+HaiCrypt Encryption/Decryption Passphrase string, length
+from 10 to 79 characters. The passphrase is the shared
+secret between the sender and the receiver. It is used
+to generate the Key Encrypting Key using PBKDF2
+(Password-Based Key Derivation Function). It is used
+only if @option{pbkeylen} is non-zero. It is used on
+the receiver only if the received data is encrypted.
+The configured passphrase cannot be recovered (write-only).
+
+@item pbkeylen=@var{bytes}
+Sender encryption key length, in bytes.
+Only can be set to 0, 16, 24 and 32.
+Enable sender encryption if not 0.
+Not required on receiver (set to 0),
+key size obtained from sender in HaiCrypt handshake.
+Default value is 0.
+
+@item recv_buffer_size=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+@item send_buffer_size=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+@item rw_timeout
+Set raise error timeout for read/write optations.
+
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
+
+@item tlpktdrop=@var{1|0}
+Too-late Packet Drop. When enabled on receiver, it skips
+missing packets that have not been delivered in time and
+delivers the following packets to the application when
+their time-to-play has come. It also sends a fake ACK to
+the sender. When enabled on sender and enabled on the
+receiving peer, the sender drops the older packets that
+have no chance of being delivered in time. It was
+automatically enabled in the sender if the receiver
+supports it.
+
+@item tsbpddelay
+Timestamp-based Packet Delivery Delay.
+Used to absorb burst of missed packet retransmission.
+
+@end table
+
+For more information see: @url{https://github.com/Haivision/srt}.
+
 @section tcp
 
 Transmission Control Protocol.
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 2c1c0f6d7f..96085d20c6 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -414,6 +414,9 @@ OBJS-$(CONFIG_TLS_PROTOCOL)              += tls.o $(TLS-OBJS-yes)
 OBJS-$(CONFIG_UDP_PROTOCOL)              += udp.o
 OBJS-$(CONFIG_UNIX_PROTOCOL)             += unix.o
 
+# external libraries
+OBJS-$(CONFIG_LIBSRT_PROTOCOL)           += libsrt.o
+
 SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
 SKIPHEADERS-$(CONFIG_NETWORK)            += network.h rtsp.h
 
diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c
new file mode 100644
index 0000000000..3e50dab64f
--- /dev/null
+++ b/libavformat/libsrt.c
@@ -0,0 +1,546 @@
+/*
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Haivision Open SRT (Secure Reliable Transport) protocol
+ */
+
+#include <srt/srt.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "libavutil/time.h"
+
+#include "avformat.h"
+#include "internal.h"
+#include "network.h"
+#include "os_support.h"
+#include "url.h"
+
+enum SRTMode {
+    SRT_MODE_CALLER = 0,
+    SRT_MODE_LISTENER = 1,
+    SRT_MODE_RENDEZVOUS = 2
+};
+
+typedef struct SRTContext {
+    const AVClass *class;
+    int fd;
+    int eid;
+    int64_t rw_timeout;
+    int64_t listen_timeout;
+    int recv_buffer_size;
+    int send_buffer_size;
+
+    int64_t maxbw;
+    int pbkeylen;
+    char *passphrase;
+    int mss;
+    int ffs;
+    int ipttl;
+    int iptos;
+    int64_t inputbw;
+    int oheadbw;
+    int64_t tsbpddelay;
+    int tlpktdrop;
+    int nakreport;
+    int64_t connect_timeout;
+    enum SRTMode mode;
+} SRTContext;
+
+#define D AV_OPT_FLAG_DECODING_PARAM
+#define E AV_OPT_FLAG_ENCODING_PARAM
+#define OFFSET(x) offsetof(SRTContext, x)
+static const AVOption libsrt_options[] = {
+    { "rw_timeout",     "Timeout of socket I/O operations",                                     OFFSET(rw_timeout),       AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "listen_timeout", "Connection awaiting timeout",                                          OFFSET(listen_timeout),   AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "send_buffer_size", "Socket send buffer size (in bytes)",                                 OFFSET(send_buffer_size), AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "recv_buffer_size", "Socket receive buffer size (in bytes)",                              OFFSET(recv_buffer_size), AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "maxbw",          "Maximum bandwidth (bytes per second) that the connection can use",     OFFSET(maxbw),            AV_OPT_TYPE_INT64,    { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "pbkeylen",       "Crypto key len in bytes {16,24,32} Default: 16 (128-bit)",             OFFSET(pbkeylen),         AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 32,        .flags = D|E },
+    { "passphrase",     "Crypto PBKDF2 Passphrase size[0,10..64] 0:disable crypto",             OFFSET(passphrase),       AV_OPT_TYPE_STRING,   { .str = NULL },              .flags = D|E },
+    { "mss",            "The Maximum Segment Size",                                             OFFSET(mss),              AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 1500,      .flags = D|E },
+    { "ffs",            "Flight flag size (window size) (in bytes)",                            OFFSET(ffs),              AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "ipttl",          "IP Time To Live",                                                      OFFSET(ipttl),            AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 255,       .flags = D|E },
+    { "iptos",          "IP Type of Service",                                                   OFFSET(iptos),            AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 255,       .flags = D|E },
+    { "inputbw",        "Estimated input stream rate",                                          OFFSET(inputbw),          AV_OPT_TYPE_INT64,    { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "oheadbw",        "MaxBW ceiling based on % over input stream rate",                      OFFSET(oheadbw),          AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 100,       .flags = D|E },
+    { "tsbpddelay",     "TsbPd receiver delay to absorb burst of missed packet retransmission", OFFSET(tsbpddelay),       AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "tlpktdrop",      "Enable receiver pkt drop",                                             OFFSET(tlpktdrop),        AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 1,         .flags = D|E },
+    { "nakreport",      "Enable receiver to send periodic NAK reports",                         OFFSET(nakreport),        AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 1,         .flags = D|E },
+    { "connect_timeout", "Connect timeout. Caller default: 3000, rendezvous (x 10)",            OFFSET(connect_timeout),  AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+    { "mode",           "Connection mode (caller, listener, rendezvous)",                       OFFSET(mode),             AV_OPT_TYPE_INT,      { .i64 = SRT_MODE_CALLER }, SRT_MODE_CALLER, SRT_MODE_RENDEZVOUS, .flags = D|E, "mode" },
+    { "caller",         NULL, 0, AV_OPT_TYPE_CONST,  { .i64 = SRT_MODE_CALLER },     INT_MIN, INT_MAX, .flags = D|E, "mode" },
+    { "listener",       NULL, 0, AV_OPT_TYPE_CONST,  { .i64 = SRT_MODE_LISTENER },   INT_MIN, INT_MAX, .flags = D|E, "mode" },
+    { "rendezvous",     NULL, 0, AV_OPT_TYPE_CONST,  { .i64 = SRT_MODE_RENDEZVOUS }, INT_MIN, INT_MAX, .flags = D|E, "mode" },
+    { NULL }
+};
+
+static int libsrt_neterrno(URLContext *h)
+{
+    int err = srt_getlasterror(NULL);
+    av_log(h, AV_LOG_ERROR, "%s\n", srt_getlasterror_str());
+    if (err == SRT_EASYNCRCV)
+        return AVERROR(EAGAIN);
+    return AVERROR_UNKNOWN;
+}
+
+static int libsrt_socket_nonblock(int socket, int enable)
+{
+    int ret = srt_setsockopt(socket, 0, SRTO_SNDSYN, &enable, sizeof(enable));
+    if (ret < 0)
+        return ret;
+    return srt_setsockopt(socket, 0, SRTO_RCVSYN, &enable, sizeof(enable));
+}
+
+static int libsrt_network_wait_fd(URLContext *h, int eid, int fd, int write)
+{
+    int ret, len = 1;
+    int modes = write ? SRT_EPOLL_OUT : SRT_EPOLL_IN;
+    SRTSOCKET ready[1];
+
+    if (srt_epoll_add_usock(eid, fd, &modes) < 0)
+        return libsrt_neterrno(h);
+    if (write) {
+        ret = srt_epoll_wait(eid, 0, 0, ready, &len, POLLING_TIME, 0, 0, 0, 0);
+    } else {
+        ret = srt_epoll_wait(eid, ready, &len, 0, 0, POLLING_TIME, 0, 0, 0, 0);
+    }
+    if (ret < 0) {
+        if (srt_getlasterror(NULL) == SRT_ETIMEOUT)
+            ret = AVERROR(EAGAIN);
+        else
+            ret = libsrt_neterrno(h);
+    } else {
+        ret = 0;
+    }
+    if (srt_epoll_remove_usock(eid, fd) < 0)
+        return libsrt_neterrno(h);
+    return ret;
+}
+
+/* TODO de-duplicate code from ff_network_wait_fd_timeout() */
+
+static int libsrt_network_wait_fd_timeout(URLContext *h, int eid, int fd, int write, int64_t timeout, AVIOInterruptCB *int_cb)
+{
+    int ret;
+    int64_t wait_start = 0;
+
+    while (1) {
+        if (ff_check_interrupt(int_cb))
+            return AVERROR_EXIT;
+        ret = libsrt_network_wait_fd(h, eid, fd, write);
+        if (ret != AVERROR(EAGAIN))
+            return ret;
+        if (timeout > 0) {
+            if (!wait_start)
+                wait_start = av_gettime_relative();
+            else if (av_gettime_relative() - wait_start > timeout)
+                return AVERROR(ETIMEDOUT);
+        }
+    }
+}
+
+static int libsrt_listen(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, URLContext *h, int timeout)
+{
+    int ret;
+    int reuse = 1;
+    if (srt_setsockopt(fd, SOL_SOCKET, SRTO_REUSEADDR, &reuse, sizeof(reuse))) {
+        av_log(h, AV_LOG_WARNING, "setsockopt(SRTO_REUSEADDR) failed\n");
+    }
+    ret = srt_bind(fd, addr, addrlen);
+    if (ret)
+        return libsrt_neterrno(h);
+
+    ret = srt_listen(fd, 1);
+    if (ret)
+        return libsrt_neterrno(h);
+
+    while ((ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback))) {
+        switch (ret) {
+        case AVERROR(ETIMEDOUT):
+            continue;
+        default:
+            return ret;
+        }
+    }
+
+    ret = srt_accept(fd, NULL, NULL);
+    if (ret < 0)
+        return libsrt_neterrno(h);
+    if (libsrt_socket_nonblock(ret, 1) < 0)
+        av_log(h, AV_LOG_DEBUG, "libsrt_socket_nonblock failed\n");
+
+    return ret;
+}
+
+static int libsrt_listen_connect(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next)
+{
+    int ret;
+
+    if (libsrt_socket_nonblock(fd, 1) < 0)
+        av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n");
+
+    while ((ret = srt_connect(fd, addr, addrlen))) {
+        ret = libsrt_neterrno(h);
+        switch (ret) {
+        case AVERROR(EINTR):
+            if (ff_check_interrupt(&h->interrupt_callback))
+                return AVERROR_EXIT;
+            continue;
+        case AVERROR(EINPROGRESS):
+        case AVERROR(EAGAIN):
+            ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback);
+            if (ret < 0)
+                return ret;
+            ret = srt_getlasterror(NULL);
+            srt_clearlasterror();
+            if (ret != 0) {
+                char buf[128];
+                ret = AVERROR(ret);
+                av_strerror(ret, buf, sizeof(buf));
+                if (will_try_next)
+                    av_log(h, AV_LOG_WARNING,
+                           "Connection to %s failed (%s), trying next address\n",
+                           h->filename, buf);
+                else
+                    av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n",
+                           h->filename, buf);
+            }
+        default:
+            return ret;
+        }
+    }
+    return ret;
+}
+
+static int libsrt_setsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, const void * optval, int optlen)
+{
+    if (srt_setsockopt(fd, 0, optname, optval, optlen) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option %s on socket: %s\n", optnamestr, srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    return 0;
+}
+
+/* - The "POST" options can be altered any time on a connected socket.
+     They MAY have also some meaning when set prior to connecting; such
+     option is SRTO_RCVSYN, which makes connect/accept call asynchronous.
+     Because of that this option is treated special way in this app. */
+static int libsrt_set_options_post(URLContext *h, int fd)
+{
+    SRTContext *s = h->priv_data;
+
+    if ((s->inputbw >= 0 && libsrt_setsockopt(h, fd, SRTO_INPUTBW, "SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) ||
+        (s->oheadbw >= 0 && libsrt_setsockopt(h, fd, SRTO_OHEADBW, "SRTO_OHEADBW", &s->oheadbw, sizeof(s->oheadbw)) < 0)) {
+        return AVERROR(EIO);
+    }
+    return 0;
+}
+
+/* - The "PRE" options must be set prior to connecting and can't be altered
+     on a connected socket, however if set on a listening socket, they are
+     derived by accept-ed socket. */
+static int libsrt_set_options_pre(URLContext *h, int fd)
+{
+    SRTContext *s = h->priv_data;
+    int yes = 1;
+    int tsbpddelay = s->tsbpddelay / 1000;
+    int connect_timeout = s->connect_timeout;
+
+    if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) ||
+        (s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) ||
+        (s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) ||
+        (s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", &s->passphrase, sizeof(s->passphrase)) < 0) ||
+        (s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS", &s->mss, sizeof(s->mss)) < 0) ||
+        (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) ||
+        (s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) ||
+        (s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) ||
+        (tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) ||
+        (s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) ||
+        (s->nakreport >= 0 && libsrt_setsockopt(h, fd, SRTO_NAKREPORT, "SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) ||
+        (connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) <0 )) {
+        return AVERROR(EIO);
+    }
+    return 0;
+}
+
+
+static int libsrt_setup(URLContext *h, const char *uri, int flags)
+{
+    struct addrinfo hints = { 0 }, *ai, *cur_ai;
+    int port, fd = -1;
+    SRTContext *s = h->priv_data;
+    const char *p;
+    char buf[256];
+    int ret;
+    char hostname[1024],proto[1024],path[1024];
+    char portstr[10];
+    int open_timeout = 5000000;
+    int eid;
+
+    eid = srt_epoll_create();
+    if (eid < 0)
+        return libsrt_neterrno(h);
+    s->eid = eid;
+
+    av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
+        &port, path, sizeof(path), uri);
+    if (strcmp(proto, "srt"))
+        return AVERROR(EINVAL);
+    if (port <= 0 || port >= 65536) {
+        av_log(h, AV_LOG_ERROR, "Port missing in uri\n");
+        return AVERROR(EINVAL);
+    }
+    p = strchr(uri, '?');
+    if (p) {
+        if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
+            s->rw_timeout = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
+            s->listen_timeout = strtol(buf, NULL, 10);
+        }
+    }
+    if (s->rw_timeout >= 0) {
+        open_timeout = h->rw_timeout = s->rw_timeout;
+    }
+    hints.ai_family = AF_UNSPEC;
+    hints.ai_socktype = SOCK_DGRAM;
+    snprintf(portstr, sizeof(portstr), "%d", port);
+    if (s->mode == SRT_MODE_LISTENER)
+        hints.ai_flags |= AI_PASSIVE;
+    ret = getaddrinfo(hostname[0] ? hostname : NULL, portstr, &hints, &ai);
+    if (ret) {
+        av_log(h, AV_LOG_ERROR,
+               "Failed to resolve hostname %s: %s\n",
+               hostname, gai_strerror(ret));
+        return AVERROR(EIO);
+    }
+
+    cur_ai = ai;
+
+ restart:
+
+    fd = srt_socket(cur_ai->ai_family, cur_ai->ai_socktype, 0);
+    if (fd < 0) {
+        ret = libsrt_neterrno(h);
+        goto fail;
+    }
+
+    if ((ret = libsrt_set_options_pre(h, fd)) < 0) {
+        goto fail;
+    }
+
+    /* Set the socket's send or receive buffer sizes, if specified.
+       If unspecified or setting fails, system default is used. */
+    if (s->recv_buffer_size > 0) {
+        srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_RCVBUF, &s->recv_buffer_size, sizeof (s->recv_buffer_size));
+    }
+    if (s->send_buffer_size > 0) {
+        srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_SNDBUF, &s->send_buffer_size, sizeof (s->send_buffer_size));
+    }
+    if (s->mode == SRT_MODE_LISTENER) {
+        // multi-client
+        if ((ret = libsrt_listen(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen, h, open_timeout / 1000)) < 0)
+            goto fail1;
+        fd = ret;
+    } else {
+        if (s->mode == SRT_MODE_RENDEZVOUS) {
+            ret = srt_bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
+            if (ret)
+                goto fail1;
+        }
+
+        if ((ret = libsrt_listen_connect(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
+                                          open_timeout / 1000, h, !!cur_ai->ai_next)) < 0) {
+            if (ret == AVERROR_EXIT)
+                goto fail1;
+            else
+                goto fail;
+        }
+    }
+    if ((ret = libsrt_set_options_post(h, fd)) < 0) {
+        goto fail;
+    }
+
+    h->is_streamed = 1;
+    s->fd = fd;
+
+    freeaddrinfo(ai);
+    return 0;
+
+ fail:
+    if (cur_ai->ai_next) {
+        /* Retry with the next sockaddr */
+        cur_ai = cur_ai->ai_next;
+        if (fd >= 0)
+            srt_close(fd);
+        ret = 0;
+        goto restart;
+    }
+ fail1:
+    if (fd >= 0)
+        srt_close(fd);
+    freeaddrinfo(ai);
+    return ret;
+}
+
+static int libsrt_open(URLContext *h, const char *uri, int flags)
+{
+    SRTContext *s = h->priv_data;
+    const char * p;
+    char buf[256];
+
+    if (srt_startup() < 0) {
+        return AVERROR_UNKNOWN;
+    }
+
+    /* SRT options (srt/srt.h) */
+    p = strchr(uri, '?');
+    if (p) {
+        if (av_find_info_tag(buf, sizeof(buf), "maxbw", p)) {
+            s->maxbw = strtoll(buf, NULL, 0);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "pbkeylen", p)) {
+            s->pbkeylen = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "passphrase", p)) {
+            s->passphrase = av_strndup(buf, strlen(buf));
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "mss", p)) {
+            s->mss = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "ffs", p)) {
+            s->ffs = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "ipttl", p)) {
+            s->ipttl = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "iptos", p)) {
+            s->iptos = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "inputbw", p)) {
+            s->inputbw = strtoll(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "oheadbw", p)) {
+            s->oheadbw = strtoll(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) {
+            s->tsbpddelay = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "tlpktdrop", p)) {
+            s->tlpktdrop = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "nakreport", p)) {
+            s->nakreport = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "connect_timeout", p)) {
+            s->connect_timeout = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "mode", p)) {
+            if (!strcmp(buf, "caller")) {
+                s->mode = SRT_MODE_CALLER;
+            } else if (!strcmp(buf, "listener")) {
+                s->mode = SRT_MODE_LISTENER;
+            } else if (!strcmp(buf, "rendezvous")) {
+                s->mode = SRT_MODE_RENDEZVOUS;
+            } else {
+                return AVERROR(EIO);
+            }
+        }
+    }
+    return libsrt_setup(h, uri, flags);
+}
+
+static int libsrt_read(URLContext *h, uint8_t *buf, int size)
+{
+    SRTContext *s = h->priv_data;
+    int ret;
+
+    if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+        ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 0, h->rw_timeout, &h->interrupt_callback);
+        if (ret)
+            return ret;
+    }
+
+    ret = srt_recvmsg(s->fd, buf, size);
+    if (ret < 0) {
+        ret = libsrt_neterrno(h);
+    }
+
+    return ret;
+}
+
+static int libsrt_write(URLContext *h, const uint8_t *buf, int size)
+{
+    SRTContext *s = h->priv_data;
+    int ret;
+
+    if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+        ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 1, h->rw_timeout, &h->interrupt_callback);
+        if (ret)
+            return ret;
+    }
+
+    ret = srt_sendmsg(s->fd, buf, size, -1, 0);
+    if (ret < 0) {
+        ret = libsrt_neterrno(h);
+    }
+
+    return ret;
+}
+
+static int libsrt_close(URLContext *h)
+{
+    SRTContext *s = h->priv_data;
+
+    srt_close(s->fd);
+
+    srt_epoll_release(s->eid);
+
+    srt_cleanup();
+
+    return 0;
+}
+
+static int libsrt_get_file_handle(URLContext *h)
+{
+    SRTContext *s = h->priv_data;
+    return s->fd;
+}
+
+static const AVClass libsrt_class = {
+    .class_name = "libsrt",
+    .item_name  = av_default_item_name,
+    .option     = libsrt_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+const URLProtocol ff_libsrt_protocol = {
+    .name                = "srt",
+    .url_open            = libsrt_open,
+    .url_read            = libsrt_read,
+    .url_write           = libsrt_write,
+    .url_close           = libsrt_close,
+    .url_get_file_handle = libsrt_get_file_handle,
+    .priv_data_size      = sizeof(SRTContext),
+    .flags               = URL_PROTOCOL_FLAG_NETWORK,
+    .priv_data_class     = &libsrt_class,
+};
diff --git a/libavformat/protocols.c b/libavformat/protocols.c
index 8ea5c0e757..15b9ed736d 100644
--- a/libavformat/protocols.c
+++ b/libavformat/protocols.c
@@ -56,6 +56,7 @@ extern const URLProtocol ff_librtmpe_protocol;
 extern const URLProtocol ff_librtmps_protocol;
 extern const URLProtocol ff_librtmpt_protocol;
 extern const URLProtocol ff_librtmpte_protocol;
+extern const URLProtocol ff_libsrt_protocol;
 
 #include "libavformat/protocol_list.c"