Merge "res_rtp_asterisk: In Developer Mode, do not require OpenSSL."

pull/11/head
Joshua Colp 7 years ago committed by Gerrit Code Review
commit 44ff1e1675

@ -278,7 +278,7 @@ struct rtp_learning_info {
enum ast_media_type stream_type; enum ast_media_type stream_type;
}; };
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
struct dtls_details { struct dtls_details {
SSL *ssl; /*!< SSL session */ SSL *ssl; /*!< SSL session */
BIO *read_bio; /*!< Memory buffer for reading */ BIO *read_bio; /*!< Memory buffer for reading */
@ -423,7 +423,7 @@ struct ast_rtp {
unsigned int ice_num_components; /*!< The number of ICE components */ unsigned int ice_num_components; /*!< The number of ICE components */
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
SSL_CTX *ssl_ctx; /*!< SSL context */ SSL_CTX *ssl_ctx; /*!< SSL context */
enum ast_rtp_dtls_verify dtls_verify; /*!< What to verify */ enum ast_rtp_dtls_verify dtls_verify; /*!< What to verify */
enum ast_srtp_suite suite; /*!< SRTP crypto suite */ enum ast_srtp_suite suite; /*!< SRTP crypto suite */
@ -500,7 +500,7 @@ struct ast_rtcp {
/* VP8: sequence number for the RTCP FIR FCI */ /* VP8: sequence number for the RTCP FIR FCI */
int firseq; int firseq;
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
struct dtls_details dtls; /*!< DTLS state information */ struct dtls_details dtls; /*!< DTLS state information */
#endif #endif
@ -568,7 +568,7 @@ static void ast_rtp_set_stream_num(struct ast_rtp_instance *instance, int stream
static int ast_rtp_extension_enable(struct ast_rtp_instance *instance, enum ast_rtp_extension extension); static int ast_rtp_extension_enable(struct ast_rtp_instance *instance, enum ast_rtp_extension extension);
static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent); static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent);
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
static int ast_rtp_activate(struct ast_rtp_instance *instance); static int ast_rtp_activate(struct ast_rtp_instance *instance);
static void dtls_srtp_check_pending(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp); static void dtls_srtp_check_pending(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp);
static void dtls_srtp_start_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp); static void dtls_srtp_start_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp);
@ -1587,7 +1587,7 @@ static struct ast_rtp_engine_ice ast_rtp_ice = {
}; };
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
static int dtls_verify_callback(int preverify_ok, X509_STORE_CTX *ctx) static int dtls_verify_callback(int preverify_ok, X509_STORE_CTX *ctx)
{ {
/* We don't want to actually verify the certificate so just accept what they have provided */ /* We don't want to actually verify the certificate so just accept what they have provided */
@ -2265,7 +2265,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
#ifdef HAVE_PJPROJECT #ifdef HAVE_PJPROJECT
.ice = &ast_rtp_ice, .ice = &ast_rtp_ice,
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
.dtls = &ast_rtp_dtls, .dtls = &ast_rtp_dtls,
.activate = ast_rtp_activate, .activate = ast_rtp_activate,
#endif #endif
@ -2277,7 +2277,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.bundle = ast_rtp_bundle, .bundle = ast_rtp_bundle,
}; };
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/*! \pre instance is locked */ /*! \pre instance is locked */
static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtls_details *dtls, int rtcp) static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtls_details *dtls, int rtcp)
{ {
@ -2311,7 +2311,7 @@ static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtl
} }
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
static void dtls_perform_setup(struct dtls_details *dtls) static void dtls_perform_setup(struct dtls_details *dtls)
{ {
if (!dtls->ssl || !SSL_is_init_finished(dtls->ssl)) { if (!dtls->ssl || !SSL_is_init_finished(dtls->ssl)) {
@ -2355,7 +2355,7 @@ static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
} }
} }
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
dtls_perform_setup(&rtp->dtls); dtls_perform_setup(&rtp->dtls);
dtls_perform_handshake(instance, &rtp->dtls, 0); dtls_perform_handshake(instance, &rtp->dtls, 0);
@ -2489,7 +2489,7 @@ static inline int rtcp_debug_test_addr(struct ast_sockaddr *addr)
return 1; return 1;
} }
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/*! \pre instance is locked */ /*! \pre instance is locked */
static int dtls_srtp_handle_timeout(struct ast_rtp_instance *instance, int rtcp) static int dtls_srtp_handle_timeout(struct ast_rtp_instance *instance, int rtcp)
{ {
@ -2835,7 +2835,7 @@ static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t s
return len; return len;
} }
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/* If this is an SSL packet pass it to OpenSSL for processing. RFC section for first byte value: /* If this is an SSL packet pass it to OpenSSL for processing. RFC section for first byte value:
* https://tools.ietf.org/html/rfc5764#section-5.1.2 */ * https://tools.ietf.org/html/rfc5764#section-5.1.2 */
if ((*in >= 20) && (*in <= 63)) { if ((*in >= 20) && (*in <= 63)) {
@ -3527,7 +3527,7 @@ static int rtp_allocate_transport(struct ast_rtp_instance *instance, struct ast_
} }
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
rtp->rekeyid = -1; rtp->rekeyid = -1;
rtp->dtls.timeout_timer = -1; rtp->dtls.timeout_timer = -1;
#endif #endif
@ -3543,7 +3543,7 @@ static void rtp_deallocate_transport(struct ast_rtp_instance *instance, struct a
struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, }; struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
ast_rtp_dtls_stop(instance); ast_rtp_dtls_stop(instance);
#endif #endif
@ -7161,7 +7161,7 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
return; return;
} }
rtp->rtcp->s = -1; rtp->rtcp->s = -1;
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
rtp->rtcp->dtls.timeout_timer = -1; rtp->rtcp->dtls.timeout_timer = -1;
#endif #endif
rtp->rtcp->schedid = -1; rtp->rtcp->schedid = -1;
@ -7224,7 +7224,7 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP); rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP);
} }
#endif #endif
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
dtls_setup_rtcp(instance); dtls_setup_rtcp(instance);
#endif #endif
} else { } else {
@ -7244,7 +7244,7 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
rtp->rtcp->s = rtp->s; rtp->rtcp->s = rtp->s;
ast_rtp_instance_get_remote_address(instance, &addr); ast_rtp_instance_get_remote_address(instance, &addr);
ast_sockaddr_copy(&rtp->rtcp->them, &addr); ast_sockaddr_copy(&rtp->rtcp->them, &addr);
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
if (rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) { if (rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) {
SSL_free(rtp->rtcp->dtls.ssl); SSL_free(rtp->rtcp->dtls.ssl);
} }
@ -7272,7 +7272,7 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) { if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) {
close(rtp->rtcp->s); close(rtp->rtcp->s);
} }
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
ao2_unlock(instance); ao2_unlock(instance);
dtls_srtp_stop_timeout_timer(instance, rtp, 1); dtls_srtp_stop_timeout_timer(instance, rtp, 1);
ao2_lock(instance); ao2_lock(instance);
@ -7547,7 +7547,7 @@ static void ast_rtp_stop(struct ast_rtp_instance *instance)
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance); struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_sockaddr addr = { {0,} }; struct ast_sockaddr addr = { {0,} };
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
ao2_unlock(instance); ao2_unlock(instance);
AST_SCHED_DEL_UNREF(rtp->sched, rtp->rekeyid, ao2_ref(instance, -1)); AST_SCHED_DEL_UNREF(rtp->sched, rtp->rekeyid, ao2_ref(instance, -1));
@ -7767,7 +7767,7 @@ static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instanc
AST_VECTOR_APPEND(&parent_rtp->ssrc_mapping, mapping); AST_VECTOR_APPEND(&parent_rtp->ssrc_mapping, mapping);
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/* If DTLS-SRTP is already in use then add the local SSRC to it, otherwise it will get added once DTLS /* If DTLS-SRTP is already in use then add the local SSRC to it, otherwise it will get added once DTLS
* negotiation has been completed. * negotiation has been completed.
*/ */
@ -7788,7 +7788,7 @@ static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instanc
return 0; return 0;
} }
#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) #if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/*! \pre instance is locked */ /*! \pre instance is locked */
static int ast_rtp_activate(struct ast_rtp_instance *instance) static int ast_rtp_activate(struct ast_rtp_instance *instance)
{ {

Loading…
Cancel
Save