blob: 5e6e5164d757fbbc1f3c144116c26d23673ec13b
1 | /* |
2 | * SRTP network protocol |
3 | * Copyright (c) 2012 Martin Storsjo |
4 | * |
5 | * This file is part of FFmpeg. |
6 | * |
7 | * FFmpeg is free software; you can redistribute it and/or |
8 | * modify it under the terms of the GNU Lesser General Public |
9 | * License as published by the Free Software Foundation; either |
10 | * version 2.1 of the License, or (at your option) any later version. |
11 | * |
12 | * FFmpeg is distributed in the hope that it will be useful, |
13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 | * Lesser General Public License for more details. |
16 | * |
17 | * You should have received a copy of the GNU Lesser General Public |
18 | * License along with FFmpeg; if not, write to the Free Software |
19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
20 | */ |
21 | |
22 | #include "libavutil/opt.h" |
23 | #include "avformat.h" |
24 | #include "avio_internal.h" |
25 | #include "url.h" |
26 | |
27 | #include "internal.h" |
28 | #include "rtpdec.h" |
29 | #include "srtp.h" |
30 | |
31 | typedef struct SRTPProtoContext { |
32 | const AVClass *class; |
33 | URLContext *rtp_hd; |
34 | const char *out_suite, *out_params; |
35 | const char *in_suite, *in_params; |
36 | struct SRTPContext srtp_out, srtp_in; |
37 | uint8_t encryptbuf[RTP_MAX_PACKET_LENGTH]; |
38 | } SRTPProtoContext; |
39 | |
40 | #define D AV_OPT_FLAG_DECODING_PARAM |
41 | #define E AV_OPT_FLAG_ENCODING_PARAM |
42 | static const AVOption options[] = { |
43 | { "srtp_out_suite", "", offsetof(SRTPProtoContext, out_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, |
44 | { "srtp_out_params", "", offsetof(SRTPProtoContext, out_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, E }, |
45 | { "srtp_in_suite", "", offsetof(SRTPProtoContext, in_suite), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D }, |
46 | { "srtp_in_params", "", offsetof(SRTPProtoContext, in_params), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, D }, |
47 | { NULL } |
48 | }; |
49 | |
50 | static const AVClass srtp_context_class = { |
51 | .class_name = "srtp", |
52 | .item_name = av_default_item_name, |
53 | .option = options, |
54 | .version = LIBAVUTIL_VERSION_INT, |
55 | }; |
56 | |
57 | static int srtp_close(URLContext *h) |
58 | { |
59 | SRTPProtoContext *s = h->priv_data; |
60 | ff_srtp_free(&s->srtp_out); |
61 | ff_srtp_free(&s->srtp_in); |
62 | ffurl_close(s->rtp_hd); |
63 | s->rtp_hd = NULL; |
64 | return 0; |
65 | } |
66 | |
67 | static int srtp_open(URLContext *h, const char *uri, int flags) |
68 | { |
69 | SRTPProtoContext *s = h->priv_data; |
70 | char hostname[256], buf[1024], path[1024]; |
71 | int rtp_port, ret; |
72 | |
73 | if (s->out_suite && s->out_params) |
74 | if ((ret = ff_srtp_set_crypto(&s->srtp_out, s->out_suite, s->out_params)) < 0) |
75 | goto fail; |
76 | if (s->in_suite && s->in_params) |
77 | if ((ret = ff_srtp_set_crypto(&s->srtp_in, s->in_suite, s->in_params)) < 0) |
78 | goto fail; |
79 | |
80 | av_url_split(NULL, 0, NULL, 0, hostname, sizeof(hostname), &rtp_port, |
81 | path, sizeof(path), uri); |
82 | ff_url_join(buf, sizeof(buf), "rtp", NULL, hostname, rtp_port, "%s", path); |
83 | if ((ret = ffurl_open_whitelist(&s->rtp_hd, buf, flags, &h->interrupt_callback, |
84 | NULL, h->protocol_whitelist, h->protocol_blacklist, h)) < 0) |
85 | goto fail; |
86 | |
87 | h->max_packet_size = FFMIN(s->rtp_hd->max_packet_size, |
88 | sizeof(s->encryptbuf)) - 14; |
89 | h->is_streamed = 1; |
90 | return 0; |
91 | |
92 | fail: |
93 | srtp_close(h); |
94 | return ret; |
95 | } |
96 | |
97 | static int srtp_read(URLContext *h, uint8_t *buf, int size) |
98 | { |
99 | SRTPProtoContext *s = h->priv_data; |
100 | int ret; |
101 | start: |
102 | ret = ffurl_read(s->rtp_hd, buf, size); |
103 | if (ret > 0 && s->srtp_in.aes) { |
104 | if (ff_srtp_decrypt(&s->srtp_in, buf, &ret) < 0) |
105 | goto start; |
106 | } |
107 | return ret; |
108 | } |
109 | |
110 | static int srtp_write(URLContext *h, const uint8_t *buf, int size) |
111 | { |
112 | SRTPProtoContext *s = h->priv_data; |
113 | if (!s->srtp_out.aes) |
114 | return ffurl_write(s->rtp_hd, buf, size); |
115 | size = ff_srtp_encrypt(&s->srtp_out, buf, size, s->encryptbuf, |
116 | sizeof(s->encryptbuf)); |
117 | if (size < 0) |
118 | return size; |
119 | return ffurl_write(s->rtp_hd, s->encryptbuf, size); |
120 | } |
121 | |
122 | static int srtp_get_file_handle(URLContext *h) |
123 | { |
124 | SRTPProtoContext *s = h->priv_data; |
125 | return ffurl_get_file_handle(s->rtp_hd); |
126 | } |
127 | |
128 | static int srtp_get_multi_file_handle(URLContext *h, int **handles, |
129 | int *numhandles) |
130 | { |
131 | SRTPProtoContext *s = h->priv_data; |
132 | return ffurl_get_multi_file_handle(s->rtp_hd, handles, numhandles); |
133 | } |
134 | |
135 | const URLProtocol ff_srtp_protocol = { |
136 | .name = "srtp", |
137 | .url_open = srtp_open, |
138 | .url_read = srtp_read, |
139 | .url_write = srtp_write, |
140 | .url_close = srtp_close, |
141 | .url_get_file_handle = srtp_get_file_handle, |
142 | .url_get_multi_file_handle = srtp_get_multi_file_handle, |
143 | .priv_data_size = sizeof(SRTPProtoContext), |
144 | .priv_data_class = &srtp_context_class, |
145 | .flags = URL_PROTOCOL_FLAG_NETWORK, |
146 | }; |
147 |