[Asterisk-code-review] res rtp asterisk: In Developer Mode, do not require OpenSSL. (asterisk[13])

Alexander Traud asteriskteam at digium.com
Fri Jul 27 08:55:16 CDT 2018


Alexander Traud has uploaded this change for review. ( https://gerrit.asterisk.org/9739


Change subject: res_rtp_asterisk: In Developer Mode, do not require OpenSSL.
......................................................................

res_rtp_asterisk: In Developer Mode, do not require OpenSSL.

OpenSSL is an optional external library and should stay optional even when
Developer Mode is configured.

ASTERISK-27990

Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b
---
M res/res_rtp_asterisk.c
1 file changed, 18 insertions(+), 18 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/39/9739/1

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index d101fcb..6f7e09e 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -270,7 +270,7 @@
 	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 {
 	SSL *ssl;         /*!< SSL session */
 	BIO *read_bio;    /*!< Memory buffer for reading */
@@ -393,7 +393,7 @@
 	unsigned int ice_num_components; /*!< The number of ICE components */
 #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 */
 	enum ast_rtp_dtls_verify dtls_verify; /*!< What to verify */
 	enum ast_srtp_suite suite;   /*!< SRTP crypto suite */
@@ -470,7 +470,7 @@
 	/* VP8: sequence number for the RTCP FIR FCI */
 	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 */
 #endif
 
@@ -524,7 +524,7 @@
 static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
 static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level);
 
-#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 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);
@@ -1543,7 +1543,7 @@
 };
 #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)
 {
 	/* We don't want to actually verify the certificate so just accept what they have provided */
@@ -1999,13 +1999,13 @@
 #ifdef HAVE_PJPROJECT
 	.ice = &ast_rtp_ice,
 #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,
 	.activate = ast_rtp_activate,
 #endif
 };
 
-#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 */
 static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtls_details *dtls, int rtcp)
 {
@@ -2066,7 +2066,7 @@
 		}
 	}
 
-#if !defined(OPENSSL_NO_SRTP) && (OPENSSL_VERSION_NUMBER >= 0x10001000L)
+#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
 	dtls_perform_handshake(instance, &rtp->dtls, 0);
 
 	if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
@@ -2197,7 +2197,7 @@
 	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 */
 static int dtls_srtp_handle_timeout(struct ast_rtp_instance *instance, int rtcp)
 {
@@ -2521,7 +2521,7 @@
 	   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:
 	 * https://tools.ietf.org/html/rfc5764#section-5.1.2 */
 	if ((*in >= 20) && (*in <= 63)) {
@@ -3225,7 +3225,7 @@
 	/* Record any information we may need */
 	rtp->sched = sched;
 
-#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->dtls.timeout_timer = -1;
 #endif
@@ -3246,7 +3246,7 @@
 	struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
 #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);
 #endif
 
@@ -5861,7 +5861,7 @@
 					return;
 				}
 				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;
 #endif
 				rtp->rtcp->schedid = -1;
@@ -5924,7 +5924,7 @@
 					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
-#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);
 #endif
 			} else {
@@ -5944,7 +5944,7 @@
 				rtp->rtcp->s = rtp->s;
 				ast_rtp_instance_get_remote_address(instance, &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) {
 					SSL_free(rtp->rtcp->dtls.ssl);
 				}
@@ -5972,7 +5972,7 @@
 				if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->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);
 				dtls_srtp_stop_timeout_timer(instance, rtp, 1);
 				ao2_lock(instance);
@@ -6234,7 +6234,7 @@
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 	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);
 	AST_SCHED_DEL_UNREF(rtp->sched, rtp->rekeyid, ao2_ref(instance, -1));
 
@@ -6329,7 +6329,7 @@
 	return res;
 }
 
-#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)
 {
 	if (!dtls->ssl || !SSL_is_init_finished(dtls->ssl)) {

-- 
To view, visit https://gerrit.asterisk.org/9739
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: newchange
Gerrit-Change-Id: Ia68a4cd5474b26d45e0f43b04032ad598022853b
Gerrit-Change-Number: 9739
Gerrit-PatchSet: 1
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
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