commit a113d231a68ff910bf24d08a9b8337d9e93ccbc0
author Philipp Hancke <philipp.hancke@googlemail.com> Mon Feb 22 14:01:42 2021
committer Commit Bot <commit-bot@chromium.org> Mon Mar 08 10:41:29 2021
tree b8c65bfa2b0c4d9c2f62595bb82b4b1ff97ece17
parent c81665cf9dd7dfe1497bea0972c7cf903363bc37

srtp: use srtp_crypto_policy_set_from_profile_for_* from libsrtp

use the helper functions
  srtp_crypto_policy_set_from_profile_for_rtp
and
  srtp_crypto_policy_set_from_profile_for_rtcp
provided by libsrtp to initialize the rtp and rtcp policies.

BUG=None

Change-Id: Ib1560c0fc1c06d9e79c1f871b028555b3b4d66d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33399}

pc/srtp_session.cc

diff --git a/pc/srtp_session.cc b/pc/srtp_session.cc
index dd3b751..8e89fdf 100644
--- a/pc/srtp_session.cc
+++ b/pc/srtp_session.cc
@@ -269,22 +269,12 @@
 
   srtp_policy_t policy;
   memset(&policy, 0, sizeof(policy));
-  if (cs == rtc::SRTP_AES128_CM_SHA1_80) {
-    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
-    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
-  } else if (cs == rtc::SRTP_AES128_CM_SHA1_32) {
-    // RTP HMAC is shortened to 32 bits, but RTCP remains 80 bits.
-    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
-    srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
-  } else if (cs == rtc::SRTP_AEAD_AES_128_GCM) {
-    srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtp);
-    srtp_crypto_policy_set_aes_gcm_128_16_auth(&policy.rtcp);
-  } else if (cs == rtc::SRTP_AEAD_AES_256_GCM) {
-    srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtp);
-    srtp_crypto_policy_set_aes_gcm_256_16_auth(&policy.rtcp);
-  } else {
-    RTC_LOG(LS_WARNING) << "Failed to " << (session_ ? "update" : "create")
-                        << " SRTP session: unsupported cipher_suite " << cs;
+  if (!(srtp_crypto_policy_set_from_profile_for_rtp(
+            &policy.rtp, (srtp_profile_t)cs) == srtp_err_status_ok &&
+        srtp_crypto_policy_set_from_profile_for_rtcp(
+            &policy.rtcp, (srtp_profile_t)cs) == srtp_err_status_ok)) {
+    RTC_LOG(LS_ERROR) << "Failed to " << (session_ ? "update" : "create")
+                      << " SRTP session: unsupported cipher_suite " << cs;
     return false;
   }