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/*
* 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 "url.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_closep(&s->rtp_hd);
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,
};
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