aboutsummaryrefslogtreecommitdiffstats
path: root/libavformat/srtpproto.c
blob: 5e6e5164d757fbbc1f3c144116c26d23673ec13b (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
/*
 * SRTP network protocol
 * Copyright (c) 2012 Martin Storsjo
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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.
 *
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "libavutil/opt.h"
#include "avformat.h"
#include "avio_internal.h"
#include "url.h"

#include "internal.h"
#include "rtpdec.h"
#include "srtp.h"

typedef struct SRTPProtoContext {
    const AVClass *class;
    URLContext *rtp_hd;
    const char *out_suite, *out_params;
    const char *in_suite, *in_params;
    struct SRTPContext srtp_out, srtp_in;
    uint8_t encryptbuf[RTP_MAX_PACKET_LENGTH];
} SRTPProtoContext;

#define D AV_OPT_FLAG_DECODING_PARAM
#define E AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
    { "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
    { "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E },
    { "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
    { "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D },
    { NULL }
};

static const AVClass srtp_context_class = {
    .class_name     = "srtp",
    .item_name      = av_default_item_name,
    .option         = options,
    .version        = LIBAVUTIL_VERSION_INT,
};

static int srtp_close(URLContext *h)
{
    SRTPProtoContext *s = h->priv_data;
    ff_srtp_free(&s->srtp_out);
    ff_srtp_free(&s->srtp_in);
    ffurl_close(s->rtp_hd);
    s->rtp_hd = NULL;
    return 0;
}

static int srtp_open(URLContext *h, const char *uri, int flags)
{
    SRTPProtoContext *s = h->priv_data;
    char hostname[256], buf[1024], path[1024];
    int rtp_port, ret;

    if (s->out_suite && s->out_params)
        if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0)
            goto fail;
    if (s->in_suite && s->in_params)
        if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0)
            goto fail;

    av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port,
                 path, sizeof(path), uri);
    ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path);
    if ((ret = ffurl_open_whitelist(&s->rtp_hd, buf, flags, &h->interrupt_callback,
                                    NULL, h->protocol_whitelist, h->protocol_blacklist, h)) < 0)
        goto fail;

    h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size,
                               sizeof(s->encryptbuf)) - 14;
    h->is_streamed = 1;
    return 0;

fail:
    srtp_close(h);
    return ret;
}

static int srtp_read(URLContext *h, uint8_t *buf, int size)
{
    SRTPProtoContext *s = h->priv_data;
    int ret;
start:
    ret = ffurl_read(s->rtp_hd, buf, size);
    if (ret > 0 && s->srtp_in.aes) {
        if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0)
            goto start;
    }
    return ret;
}

static int srtp_write(URLContext *h, const uint8_t *buf, int size)
{
    SRTPProtoContext *s = h->priv_data;
    if (!s->srtp_out.aes)
        return ffurl_write(s->rtp_hd, buf, size);
    size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf,
                           sizeof(s->encryptbuf));
    if (size < 0)
        return size;
    return ffurl_write(s->rtp_hd, s->encryptbuf, size);
}

static int srtp_get_file_handle(URLContext *h)
{
    SRTPProtoContext *s = h->priv_data;
    return ffurl_get_file_handle(s->rtp_hd);
}

static int srtp_get_multi_file_handle(URLContext *h, int **handles,
                                      int *numhandles)
{
    SRTPProtoContext *s = h->priv_data;
    return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles);
}

const URLProtocol ff_srtp_protocol = {
    .name                      = "srtp",
    .url_open                  = srtp_open,
    .url_read                  = srtp_read,
    .url_write                 = srtp_write,
    .url_close                 = srtp_close,
    .url_get_file_handle       = srtp_get_file_handle,
    .url_get_multi_file_handle = srtp_get_multi_file_handle,
    .priv_data_size            = sizeof(SRTPProtoContext),
    .priv_data_class           = &srtp_context_class,
    .flags                     = URL_PROTOCOL_FLAG_NETWORK,
};