源码下载在github采用的库为libsrtp2.5.0:
weget https://github.com/cisco/libsrtp/archive/refs/tags/v2.5.0.tar.gz
新增交叉编译脚本,这里需要支持openssl。
./configure --host=arm-linux-androideabi --prefix=$(pwd)/object --enable-openssl crypto_LIBS="-L$(pwd)/../../third_party/openssl-1.1.0h/lib" crypto_CFLAGS="-I$(pwd)/../../third_party/openssl-1.1.0h/include" && make && make install
下面是交叉编译过程
tar -xvf libsrtp-2.5.0.tar.gz
cd libsrtp-2.5.0
touch config_android.sh
./config_android.sh
./
API设计采用handle的实现方式,init只需要进程初始化一次。srtp_app_alloc需要每次会话创建都进行一次初始化,需要传入srtp秘钥。protect和unprotect是实时进行加解密操作。
//srtp handle
typedef struct srtp_handle_t{
int (*protect)(void *pthis, void *rtp, size_t *pkt_octet_len);
int (*unprotect)(void *pthis, void *audio_dat, size_t *pkt_octet_len);
void* priv;
}srtp_handle_t;
#ifdef __cplusplus
extern "C" {
#endif
// 初始化
int srtp_app_init();
void srtp_app_denit();
//会话创建
int srtp_app_alloc(srtp_handle_t **pthis, const char *srtp_dec, const char *srtp_enc);
void srtp_app_free(srtp_handle_t *pthis);
初始化和反初始化,只需要在开启进程初始化一次。
int srtp_app_init()
{
int ret = srtp_init();
if(ret < 0)
return -1;
return 0;
}
void srtp_app_denit()
{
srtp_shutdown();
}
会话创建和销毁,需要每次建立会话都要重新创建,每次的秘钥都需要采用协商的加密和解密秘钥。
static int srtp_app_set_crypto_suites(int profile, srtp_crypto_policy_t *rtp)
{
if(rtp == NULL)
{
return -1;
}
switch (profile)
{
case rl_srtp_profile_aes128_cm_sha1_32:
{
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(rtp);
rl_log_debug("loc srtp_profile_aes128_cm_sha1_32");
break;
}
case rl_srtp_profile_aes128_cm_sha1_80:
{
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(rtp);
rl_log_debug("loc srtp_profile_aes128_cm_sha1_80");
break;
}
case rl_srtp_profile_aes256_cm_sha1_32:
{
srtp_crypto_policy_set_aes_cm_256_hmac_sha1_32(rtp);
rl_log_debug("loc srtp_profile_aes256_cm_sha1_32");
break;
}
case rl_srtp_profile_aes256_cm_sha1_80:
{
srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(rtp);
rl_log_debug("loc srtp_profile_aes256_cm_sha1_80");
break;
}
default:
{
rl_log_debug("loc policy none");
break;
}
}
return 0;
}
static int srtp_app_start(void *pthis, const char *srtp_dec, const char *strp_enc)
{
rl_log_debug("SRTP start");
srtp_config_t srtp_config_loc;
srtp_config_t srtp_config_dist;
srtp_handle_t *p = (srtp_handle_t *)pthis;
if((!pthis || !srtp_dec || !strp_enc || strlen(srtp_dec) == 0 || strlen(strp_enc) == 0))
return -1;
srtp_handle_priv_t* priv = p->priv;
if (!priv)
return -1;
int ret = srtp_parse_config(strp_enc, &srtp_config_loc);
if ( ret < 0 )
{
return ret;
}
ret = srtp_parse_config(srtp_dec, &srtp_config_dist);
if ( ret < 0 )
{
return ret;
}
priv->srtp_opt = 1;
memcpy(priv->srtp_params.key_loc, srtp_config_loc.key, 30);
memcpy(priv->srtp_params.key_dist, srtp_config_dist.key, 30);
srtp_app_set_crypto_suites(srtp_config_loc.profile, &priv->srtp_params.policy_loc.rtp);
priv->srtp_params.policy_loc.key = priv->srtp_params.key_loc;
priv->srtp_params.policy_loc.next = NULL;
srtp_app_set_crypto_suites(srtp_config_dist.profile, &priv->srtp_params.policy_dist.rtp);
priv->srtp_params.policy_dist.ssrc.type = ssrc_any_inbound;
priv->srtp_params.policy_dist.key = priv->srtp_params.key_dist;
priv->srtp_params.policy_dist.next = NULL;
rl_log_debug("srtp_create ret=%i", srtp_create(&priv->srtp_params.scall_session,
&priv->srtp_params.policy_dist));
return 0;
}
static void srtp_app_stop(void *pthis)
{
srtp_handle_t *p = (srtp_handle_t *)pthis;
if(!pthis)
return;
srtp_handle_priv_t* priv = p->priv;
if (!priv)
return;
rl_log_debug("SRTP stop\n");
if (priv->srtp_opt > 0)
{
priv->srtp_opt = 0;
priv->srtp_params.prev_ssrc = 0;
rl_memset(priv->srtp_params.key_loc, 0, sizeof(priv->srtp_params.key_loc));
rl_memset(priv->srtp_params.key_dist, 0, sizeof(priv->srtp_params.key_dist));
rl_log_debug("srtp_dealloc ret=%i", srtp_dealloc(priv->srtp_params.scall_session));
}
}
int srtp_app_alloc(srtp_handle_t **pthis, const char *srtp_dec, const char *srtp_enc)
{
if(!pthis)
return -1;
srtp_handle_t* p = (srtp_handle_t*)malloc(sizeof(srtp_handle_t));
if (p == NULL)
{
rl_log_err("[%s-%d] malloc failed.",__FUNCTION__, __LINE__);
return -1;
}
memset(p, 0, sizeof(srtp_handle_t));
p->protect = srtp_app_protect;
p->unprotect = srtp_app_unprotect;
p->priv = (srtp_handle_priv_t *)malloc(sizeof(srtp_handle_priv_t));
if(!p->priv)
{
free(p);
return -1;
}
memset(p->priv , 0, sizeof(srtp_handle_priv_t));
int ret = srtp_app_start(p, srtp_dec, srtp_enc);
if (ret != 0)
{
srtp_app_free(p);
return -1;
}
*pthis = p;
return 0;
}
void srtp_app_free(srtp_handle_t *pthis)
{
if(!pthis)
return;
srtp_app_stop(pthis);
if (pthis->priv){
free(pthis->priv);
pthis->priv = NULL;
}
free(pthis);
}
加密和解密只需要传入rtp封装后的数据和长度即可进行加解密操作,操作后数据是直接替换原数据的。
static int srtp_app_protect(void *pthis, void *rtp, size_t *pkt_octet_len)
{
if(!pthis || !rtp || !pkt_octet_len)
return -1;
int ret = 0;
unsigned int current_ssrc;
srtp_handle_t *p = (srtp_handle_t *)pthis;
srtp_handle_priv_t* priv = p->priv;
if (!priv)
return -1;
srtp_t session = priv->srtp_params.scall_session;
if(priv->srtp_opt)
{
current_ssrc = *((unsigned int *)((char *)rtp + 8));
current_ssrc = htonl(current_ssrc);
if(priv->srtp_params.prev_ssrc != current_ssrc)
{
if (priv->srtp_params.prev_ssrc > 0 )
{
srtp_app_remove_stream(priv->srtp_params.scall_session, priv->srtp_params.prev_ssrc);
}
priv->srtp_params.policy_loc.ssrc.type = ssrc_specific;
priv->srtp_params.policy_loc.ssrc.value = current_ssrc;
rl_log_debug("new ssrc 0x%x, %u", current_ssrc,current_ssrc);
srtp_app_add_stream(priv->srtp_params.scall_session, &priv->srtp_params.policy_loc);
priv->srtp_params.prev_ssrc = current_ssrc;
}
ret = srtp_protect(priv->srtp_params.scall_session, rtp, (int *)pkt_octet_len);
if(ret > 0)
{
rl_log_err("--srtp_protect problem ret=%i", ret);
return -1;
}
}
return 0;
}
static int srtp_app_unprotect(void *pthis, void *audio_dat, size_t *pkt_octet_len)
{
if(!pthis || !audio_dat || !pkt_octet_len)
return -1;
int ret = 0;
srtp_handle_t *p = (srtp_handle_t *)pthis;
srtp_handle_priv_t* priv = p->priv;
if (!priv)
return -1;
srtp_t session = priv->srtp_params.scall_session;
if(priv->srtp_opt)
{
ret = srtp_unprotect(priv->srtp_params.scall_session, audio_dat, (int *)pkt_octet_len);
if(ret > 0)
{
/* If the decryption fail then d'nt give this packet to DSP throw the packets */
rl_log_err("rtp_SRtpPlRecv srtp unencryption problems ret=%i", ret);
return -1;
}
}
return 0;
}
SRTP加密后可以通过wareshark抓包看到协议部分是显示SRTP,具体如下图所示。
其实wareshark是通过sip协商RTP/SAVP来显示srtp的标识,所以SDP协商一定要正确。