diff --git a/doc/protocols.texi b/doc/protocols.texi
index b34f29eebf..fb7725e058 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -1306,10 +1306,10 @@ set by the peer side. Before version 1.3.0 this option
 is only available as @option{latency}.
 
 @item recv_buffer_size=@var{bytes}
-Set receive buffer size, expressed in bytes.
+Set UDP receive buffer size, expressed in bytes.
 
 @item send_buffer_size=@var{bytes}
-Set send buffer size, expressed in bytes.
+Set UDP send buffer size, expressed in bytes.
 
 @item rw_timeout
 Set raise error timeout for read/write optations.
@@ -1329,6 +1329,87 @@ have no chance of being delivered in time. It was
 automatically enabled in the sender if the receiver
 supports it.
 
+@item sndbuf=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+@item rcvbuf=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+Receive buffer must not be greater than @option{ffs}.
+
+@item lossmaxttl=@var{packets}
+The value up to which the Reorder Tolerance may grow. When
+Reorder Tolerance is > 0, then packet loss report is delayed
+until that number of packets come in. Reorder Tolerance
+increases every time a "belated" packet has come, but it
+wasn't due to retransmission (that is, when UDP packets tend
+to come out of order), with the difference between the latest
+sequence and this packet's sequence, and not more than the
+value of this option. By default it's 0, which means that this
+mechanism is turned off, and the loss report is always sent
+immediately upon experiencing a "gap" in sequences.
+
+@item minversion
+The minimum SRT version that is required from the peer. A connection
+to a peer that does not satisfy the minimum version requirement
+will be rejected.
+
+The version format in hex is 0xXXYYZZ for x.y.z in human readable
+form.
+
+@item streamid=@var{string}
+A string limited to 512 characters that can be set on the socket prior
+to connecting. This stream ID will be able to be retrieved by the
+listener side from the socket that is returned from srt_accept and
+was connected by a socket with that set stream ID. SRT does not enforce
+any special interpretation of the contents of this string.
+This option doesn’t make sense in Rendezvous connection; the result
+might be that simply one side will override the value from the other
+side and it’s the matter of luck which one would win
+
+@item smoother=@var{live|file}
+The type of Smoother used for the transmission for that socket, which
+is responsible for the transmission and congestion control. The Smoother
+type must be exactly the same on both connecting parties, otherwise
+the connection is rejected.
+
+@item messageapi=@var{1|0}
+When set, this socket uses the Message API, otherwise it uses Buffer
+API. Note that in live mode (see @option{transtype}) there’s only
+message API available. In File mode you can chose to use one of two modes:
+
+Stream API (default, when this option is false). In this mode you may
+send as many data as you wish with one sending instruction, or even use
+dedicated functions that read directly from a file. The internal facility
+will take care of any speed and congestion control. When receiving, you
+can also receive as many data as desired, the data not extracted will be
+waiting for the next call. There is no boundary between data portions in
+the Stream mode.
+
+Message API. In this mode your single sending instruction passes exactly
+one piece of data that has boundaries (a message). Contrary to Live mode,
+this message may span across multiple UDP packets and the only size
+limitation is that it shall fit as a whole in the sending buffer. The
+receiver shall use as large buffer as necessary to receive the message,
+otherwise the message will not be given up. When the message is not
+complete (not all packets received or there was a packet loss) it will
+not be given up.
+
+@item transtype=@var{live|file}
+Sets the transmission type for the socket, in particular, setting this
+option sets multiple other parameters to their default values as required
+for a particular transmission type.
+
+live: Set options as for live transmission. In this mode, you should
+send by one sending instruction only so many data that fit in one UDP packet,
+and limited to the value defined first in @option{payload_size} (1316 is
+default in this mode). There is no speed control in this mode, only the
+bandwidth control, if configured, in order to not exceed the bandwidth with
+the overhead transmission (retransmitted and control packets).
+
+file: Set options as for non-live transmission. See @option{messageapi}
+for further explanations
+
 @end table
 
 For more information see: @url{https://github.com/Haivision/srt}.
diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c
index fbfd6ace83..fe3b312151 100644
--- a/libavformat/libsrt.c
+++ b/libavformat/libsrt.c
@@ -76,6 +76,14 @@ typedef struct SRTContext {
     int64_t rcvlatency;
     int64_t peerlatency;
     enum SRTMode mode;
+    int sndbuf;
+    int rcvbuf;
+    int lossmaxttl;
+    int minversion;
+    char *streamid;
+    char *smoother;
+    int messageapi;
+    SRT_TRANSTYPE transtype;
 } SRTContext;
 
 #define D AV_OPT_FLAG_DECODING_PARAM
@@ -110,6 +118,16 @@ static const AVOption libsrt_options[] = {
     { "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" },
+    { "sndbuf",         "Send buffer size (in bytes)",                                          OFFSET(sndbuf),           AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "rcvbuf",         "Receive buffer size (in bytes)",                                       OFFSET(rcvbuf),           AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "lossmaxttl",     "Maximum possible packet reorder tolerance",                            OFFSET(lossmaxttl),       AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "minversion",     "The minimum SRT version that is required from the peer",               OFFSET(minversion),       AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "streamid",       "A string of up to 512 characters that an Initiator can pass to a Responder",  OFFSET(streamid),  AV_OPT_TYPE_STRING,   { .str = NULL },              .flags = D|E },
+    { "smoother",       "The type of Smoother used for the transmission for that socket",       OFFSET(smoother),         AV_OPT_TYPE_STRING,   { .str = NULL },              .flags = D|E },
+    { "messageapi",     "Enable message API",                                                   OFFSET(messageapi),       AV_OPT_TYPE_INT,      { .i64 = -1 }, -1, 1,         .flags = D|E },
+    { "transtype",      "The transmission type for the socket",                                 OFFSET(transtype),        AV_OPT_TYPE_INT,      { .i64 = SRTT_INVALID }, SRTT_LIVE, SRTT_INVALID, .flags = D|E, "transtype" },
+    { "live",           NULL, 0, AV_OPT_TYPE_CONST,  { .i64 = SRTT_LIVE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" },
+    { "file",           NULL, 0, AV_OPT_TYPE_CONST,  { .i64 = SRTT_FILE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" },
     { NULL }
 };
 
@@ -297,6 +315,7 @@ static int libsrt_set_options_pre(URLContext *h, int fd)
     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->transtype != SRTT_INVALID && libsrt_setsockopt(h, fd, SRTO_TRANSTYPE, "SRTO_TRANSTYPE", &s->transtype, sizeof(s->transtype)) < 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, strlen(s->passphrase)) < 0) ||
@@ -310,6 +329,13 @@ static int libsrt_set_options_pre(URLContext *h, int fd)
         (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 ) ||
+        (s->sndbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_SNDBUF, "SRTO_SNDBUF", &s->sndbuf, sizeof(s->sndbuf)) < 0) ||
+        (s->rcvbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVBUF, "SRTO_RCVBUF", &s->rcvbuf, sizeof(s->rcvbuf)) < 0) ||
+        (s->lossmaxttl >= 0 && libsrt_setsockopt(h, fd, SRTO_LOSSMAXTTL, "SRTO_LOSSMAXTTL", &s->lossmaxttl, sizeof(s->lossmaxttl)) < 0) ||
+        (s->minversion >= 0 && libsrt_setsockopt(h, fd, SRTO_MINVERSION, "SRTO_MINVERSION", &s->minversion, sizeof(s->minversion)) < 0) ||
+        (s->streamid && libsrt_setsockopt(h, fd, SRTO_STREAMID, "SRTO_STREAMID", s->streamid, strlen(s->streamid)) < 0) ||
+        (s->smoother && libsrt_setsockopt(h, fd, SRTO_SMOOTHER, "SRTO_SMOOTHER", s->smoother, strlen(s->smoother)) < 0) ||
+        (s->messageapi >= 0 && libsrt_setsockopt(h, fd, SRTO_MESSAGEAPI, "SRTO_MESSAGEAPI", &s->messageapi, sizeof(s->messageapi)) < 0) ||
         (s->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0)) {
         return AVERROR(EIO);
     }
@@ -522,6 +548,38 @@ static int libsrt_open(URLContext *h, const char *uri, int flags)
                 return AVERROR(EIO);
             }
         }
+        if (av_find_info_tag(buf, sizeof(buf), "sndbuf", p)) {
+            s->sndbuf = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "rcvbuf", p)) {
+            s->rcvbuf = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "lossmaxttl", p)) {
+            s->lossmaxttl = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "minversion", p)) {
+            s->minversion = strtol(buf, NULL, 0);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "streamid", p)) {
+            av_freep(&s->streamid);
+            s->streamid = av_strdup(buf);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) {
+            av_freep(&s->smoother);
+            s->smoother = av_strdup(buf);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "messageapi", p)) {
+            s->messageapi = strtol(buf, NULL, 10);
+        }
+        if (av_find_info_tag(buf, sizeof(buf), "transtype", p)) {
+            if (!strcmp(buf, "live")) {
+                s->transtype = SRTT_LIVE;
+            } else if (!strcmp(buf, "file")) {
+                s->transtype = SRTT_FILE;
+            } else {
+                return AVERROR(EINVAL);
+            }
+        }
     }
     return libsrt_setup(h, uri, flags);
 }
