[Pkg-voip-commits] [janus] 12/37: Drop patches cherry-picked upstream and since included.
Jonas Smedegaard
dr at jones.dk
Mon Mar 13 19:14:32 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a commit to branch master
in repository janus.
commit 22495f9e717215533172856a7bd57b1d30189edf
Author: Jonas Smedegaard <dr at jones.dk>
Date: Sun Mar 12 14:22:25 2017 +0100
Drop patches cherry-picked upstream and since included.
---
debian/patches/020161215~871c362.patch | 257 ---------------------------------
debian/patches/020161215~aa00c95.patch | 17 ---
debian/patches/series | 2 -
3 files changed, 276 deletions(-)
diff --git a/debian/patches/020161215~871c362.patch b/debian/patches/020161215~871c362.patch
deleted file mode 100644
index b49d0a3..0000000
--- a/debian/patches/020161215~871c362.patch
+++ /dev/null
@@ -1,257 +0,0 @@
-Description: Add support for libsrtp2 to SIP plugin
-Origin: upstream, https://github.com/meetecho/janus-gateway/commit/871c362
-Author: Lorenzo Miniero <lminiero at gmail.com>
-Forwarded: yes
-Bug: https://github.com/meetecho/janus-gateway/issues/709
-Last-Update: 2017-02-05
-
---- a/plugins/janus_sip.c
-+++ b/plugins/janus_sip.c
-@@ -62,9 +62,6 @@
- #include <sofia-sip/url.h>
- #include <sofia-sip/tport_tag.h>
-
--#include <srtp/srtp.h>
--#include <srtp/crypto_kernel.h>
--
- #include "../debug.h"
- #include "../apierror.h"
- #include "../config.h"
-@@ -75,6 +72,35 @@
- #include "../sdp-utils.h"
- #include "../utils.h"
-
-+#ifdef HAVE_SRTP_2
-+#include <srtp2/srtp.h>
-+#include <openssl/rand.h>
-+#include <openssl/err.h>
-+static int srtp_crypto_get_random(uint8_t *key, int len) {
-+ /* libsrtp 2.0 doesn't have crypto_get_random, we use OpenSSL's RAND_* to replace it:
-+ * https://wiki.openssl.org/index.php/Random_Numbers */
-+ int rc = RAND_bytes(key, len);
-+ if(rc != 1) {
-+ /* Error generating */
-+ JANUS_LOG(LOG_ERR, "RAND_bytes failes: %s\n", ERR_reason_error_string(ERR_get_error()));
-+ return -1;
-+ }
-+ return 0;
-+}
-+#else
-+#include <srtp/srtp.h>
-+#include <srtp/crypto_kernel.h>
-+#define srtp_err_status_t err_status_t
-+#define srtp_err_status_ok err_status_ok
-+#define srtp_err_status_replay_fail err_status_replay_fail
-+#define srtp_err_status_replay_old err_status_replay_old
-+#define srtp_crypto_policy_set_rtp_default crypto_policy_set_rtp_default
-+#define srtp_crypto_policy_set_rtcp_default crypto_policy_set_rtcp_default
-+#define srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32 crypto_policy_set_aes_cm_128_hmac_sha1_32
-+#define srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80 crypto_policy_set_aes_cm_128_hmac_sha1_80
-+#define srtp_crypto_get_random crypto_get_random
-+#endif
-+
-
- /* Plugin information */
- #define JANUS_SIP_VERSION 6
-@@ -379,31 +405,31 @@
- #define SRTP_MASTER_LENGTH (SRTP_MASTER_KEY_LENGTH + SRTP_MASTER_SALT_LENGTH)
- static const char *janus_sip_srtp_error[] =
- {
-- "err_status_ok",
-- "err_status_fail",
-- "err_status_bad_param",
-- "err_status_alloc_fail",
-- "err_status_dealloc_fail",
-- "err_status_init_fail",
-- "err_status_terminus",
-- "err_status_auth_fail",
-- "err_status_cipher_fail",
-- "err_status_replay_fail",
-- "err_status_replay_old",
-- "err_status_algo_fail",
-- "err_status_no_such_op",
-- "err_status_no_ctx",
-- "err_status_cant_check",
-- "err_status_key_expired",
-- "err_status_socket_err",
-- "err_status_signal_err",
-- "err_status_nonce_bad",
-- "err_status_read_fail",
-- "err_status_write_fail",
-- "err_status_parse_err",
-- "err_status_encode_err",
-- "err_status_semaphore_err",
-- "err_status_pfkey_err",
-+ "srtp_err_status_ok",
-+ "srtp_err_status_fail",
-+ "srtp_err_status_bad_param",
-+ "srtp_err_status_alloc_fail",
-+ "srtp_err_status_dealloc_fail",
-+ "srtp_err_status_init_fail",
-+ "srtp_err_status_terminus",
-+ "srtp_err_status_auth_fail",
-+ "srtp_err_status_cipher_fail",
-+ "srtp_err_status_replay_fail",
-+ "srtp_err_status_replay_old",
-+ "srtp_err_status_algo_fail",
-+ "srtp_err_status_no_such_op",
-+ "srtp_err_status_no_ctx",
-+ "srtp_err_status_cant_check",
-+ "srtp_err_status_key_expired",
-+ "srtp_err_status_socket_err",
-+ "srtp_err_status_signal_err",
-+ "srtp_err_status_nonce_bad",
-+ "srtp_err_status_read_fail",
-+ "srtp_err_status_write_fail",
-+ "srtp_err_status_parse_err",
-+ "srtp_err_status_encode_err",
-+ "srtp_err_status_semaphore_err",
-+ "srtp_err_status_pfkey_err",
- };
- static const gchar *janus_sip_get_srtp_error(int error) {
- if(error < 0 || error > 24)
-@@ -415,17 +441,17 @@
- return -1;
- /* Generate key/salt */
- uint8_t *key = g_malloc0(SRTP_MASTER_LENGTH);
-- crypto_get_random(key, SRTP_MASTER_LENGTH);
-+ srtp_crypto_get_random(key, SRTP_MASTER_LENGTH);
- /* Set SRTP policies */
- srtp_policy_t *policy = video ? &session->media.video_local_policy : &session->media.audio_local_policy;
-- crypto_policy_set_rtp_default(&(policy->rtp));
-- crypto_policy_set_rtcp_default(&(policy->rtcp));
-+ srtp_crypto_policy_set_rtp_default(&(policy->rtp));
-+ srtp_crypto_policy_set_rtcp_default(&(policy->rtcp));
- policy->ssrc.type = ssrc_any_inbound;
- policy->key = key;
- policy->next = NULL;
- /* Create SRTP context */
-- err_status_t res = srtp_create(video ? &session->media.video_srtp_out : &session->media.audio_srtp_out, policy);
-- if(res != err_status_ok) {
-+ srtp_err_status_t res = srtp_create(video ? &session->media.video_srtp_out : &session->media.audio_srtp_out, policy);
-+ if(res != srtp_err_status_ok) {
- /* Something went wrong... */
- JANUS_LOG(LOG_ERR, "Oops, error creating outbound SRTP session: %d (%s)\n", res, janus_sip_get_srtp_error(res));
- g_free(key);
-@@ -452,21 +478,21 @@
- }
- /* Set SRTP policies */
- srtp_policy_t *policy = video ? &session->media.video_remote_policy : &session->media.audio_remote_policy;
-- crypto_policy_set_rtp_default(&(policy->rtp));
-- crypto_policy_set_rtcp_default(&(policy->rtcp));
-+ srtp_crypto_policy_set_rtp_default(&(policy->rtp));
-+ srtp_crypto_policy_set_rtcp_default(&(policy->rtcp));
- if(suite == 32) {
-- crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtp));
-- crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtcp));
-+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtp));
-+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&(policy->rtcp));
- } else if(suite == 80) {
-- crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtp));
-- crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtcp));
-+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtp));
-+ srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&(policy->rtcp));
- }
- policy->ssrc.type = ssrc_any_inbound;
- policy->key = decoded;
- policy->next = NULL;
- /* Create SRTP context */
-- err_status_t res = srtp_create(video ? &session->media.video_srtp_in : &session->media.audio_srtp_in, policy);
-- if(res != err_status_ok) {
-+ srtp_err_status_t res = srtp_create(video ? &session->media.video_srtp_in : &session->media.audio_srtp_in, policy);
-+ if(res != srtp_err_status_ok) {
- /* Something went wrong... */
- JANUS_LOG(LOG_ERR, "Oops, error creating inbound SRTP session: %d (%s)\n", res, janus_sip_get_srtp_error(res));
- g_free(decoded);
-@@ -772,6 +798,11 @@
- janus_config_destroy(config);
- config = NULL;
-
-+#ifdef HAVE_SRTP_2
-+ /* Init randomizer (for randum numbers in SRTP) */
-+ RAND_poll();
-+#endif
-+
- /* Setup sofia */
- su_init();
-
-@@ -1080,7 +1111,7 @@
- memcpy(&sbuf, buf, len);
- int protected = len;
- int res = srtp_protect(session->media.video_srtp_out, &sbuf, &protected);
-- if(res != err_status_ok) {
-+ if(res != srtp_err_status_ok) {
- rtp_header *header = (rtp_header *)&sbuf;
- guint32 timestamp = ntohl(header->timestamp);
- guint16 seq = ntohs(header->seq_number);
-@@ -1114,7 +1145,7 @@
- memcpy(&sbuf, buf, len);
- int protected = len;
- int res = srtp_protect(session->media.audio_srtp_out, &sbuf, &protected);
-- if(res != err_status_ok) {
-+ if(res != srtp_err_status_ok) {
- rtp_header *header = (rtp_header *)&sbuf;
- guint32 timestamp = ntohl(header->timestamp);
- guint16 seq = ntohs(header->seq_number);
-@@ -1157,7 +1188,7 @@
- memcpy(&sbuf, buf, len);
- int protected = len;
- int res = srtp_protect_rtcp(session->media.video_srtp_out, &sbuf, &protected);
-- if(res != err_status_ok) {
-+ if(res != srtp_err_status_ok) {
- JANUS_LOG(LOG_ERR, "[SIP-%s] Video SRTCP protect error... %s (len=%d-->%d)...\n",
- session->account.username, janus_sip_get_srtp_error(res), len, protected);
- } else {
-@@ -1181,7 +1212,7 @@
- memcpy(&sbuf, buf, len);
- int protected = len;
- int res = srtp_protect_rtcp(session->media.audio_srtp_out, &sbuf, &protected);
-- if(res != err_status_ok) {
-+ if(res != srtp_err_status_ok) {
- JANUS_LOG(LOG_ERR, "[SIP-%s] Audio SRTCP protect error... %s (len=%d-->%d)...\n",
- session->account.username, janus_sip_get_srtp_error(res), len, protected);
- } else {
-@@ -3103,8 +3134,8 @@
- /* Is this SRTP? */
- if(session->media.has_srtp_remote) {
- int buflen = bytes;
-- err_status_t res = srtp_unprotect(session->media.audio_srtp_in, buffer, &buflen);
-- if(res != err_status_ok && res != err_status_replay_fail && res != err_status_replay_old) {
-+ srtp_err_status_t res = srtp_unprotect(session->media.audio_srtp_in, buffer, &buflen);
-+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
- guint32 timestamp = ntohl(header->timestamp);
- guint16 seq = ntohs(header->seq_number);
- JANUS_LOG(LOG_ERR, "[SIP-%s] Audio SRTP unprotect error: %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")\n",
-@@ -3153,8 +3184,8 @@
- /* Is this SRTCP? */
- if(session->media.has_srtp_remote) {
- int buflen = bytes;
-- err_status_t res = srtp_unprotect_rtcp(session->media.audio_srtp_in, buffer, &buflen);
-- if(res != err_status_ok && res != err_status_replay_fail && res != err_status_replay_old) {
-+ srtp_err_status_t res = srtp_unprotect_rtcp(session->media.audio_srtp_in, buffer, &buflen);
-+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
- JANUS_LOG(LOG_ERR, "[SIP-%s] Audio SRTCP unprotect error: %s (len=%d-->%d)\n",
- session->account.username, janus_sip_get_srtp_error(res), bytes, buflen);
- continue;
-@@ -3180,8 +3211,8 @@
- /* Is this SRTP? */
- if(session->media.has_srtp_remote) {
- int buflen = bytes;
-- err_status_t res = srtp_unprotect(session->media.video_srtp_in, buffer, &buflen);
-- if(res != err_status_ok && res != err_status_replay_fail && res != err_status_replay_old) {
-+ srtp_err_status_t res = srtp_unprotect(session->media.video_srtp_in, buffer, &buflen);
-+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
- guint32 timestamp = ntohl(header->timestamp);
- guint16 seq = ntohs(header->seq_number);
- JANUS_LOG(LOG_ERR, "[SIP-%s] Video SRTP unprotect error: %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")\n",
-@@ -3229,8 +3260,8 @@
- /* Is this SRTCP? */
- if(session->media.has_srtp_remote) {
- int buflen = bytes;
-- err_status_t res = srtp_unprotect_rtcp(session->media.video_srtp_in, buffer, &buflen);
-- if(res != err_status_ok && res != err_status_replay_fail && res != err_status_replay_old) {
-+ srtp_err_status_t res = srtp_unprotect_rtcp(session->media.video_srtp_in, buffer, &buflen);
-+ if(res != srtp_err_status_ok && res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
- JANUS_LOG(LOG_ERR, "[SIP-%s] Video SRTP unprotect error: %s (len=%d-->%d)\n",
- session->account.username, janus_sip_get_srtp_error(res), bytes, buflen);
- continue;
diff --git a/debian/patches/020161215~aa00c95.patch b/debian/patches/020161215~aa00c95.patch
deleted file mode 100644
index d1c2945..0000000
--- a/debian/patches/020161215~aa00c95.patch
+++ /dev/null
@@ -1,17 +0,0 @@
-Description: Add support for libsrtp2 to SIP plugin
-Origin: upstream, https://github.com/meetecho/janus-gateway/commit/aa00c95
-Author: Lorenzo Miniero <lminiero at gmail.com>
-Forwarded: yes
-Bug: https://github.com/meetecho/janus-gateway/issues/709
-Last-Update: 2017-02-05
-
---- a/Makefile.am
-+++ b/Makefile.am
-@@ -126,7 +126,6 @@
- $(BORINGSSL_LIBS) \
- $(JANUS_LIBS) \
- $(JANUS_MANUAL_LIBS) \
-- -lsrtp \
- $(NULL)
-
- BUILT_SOURCES = cmdline.c cmdline.h
diff --git a/debian/patches/series b/debian/patches/series
index f038226..cf725dd 100644
--- a/debian/patches/series
+++ b/debian/patches/series
@@ -1,5 +1,3 @@
-020161215~871c362.patch
-020161215~aa00c95.patch
0001-disable-demos.patch
0002-fix-typo.patch
0003-manpages.patch
--
Alioth's /usr/local/bin/git-commit-notice on /srv/git.debian.org/git/pkg-voip/janus.git
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