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srtp: document rationale for srtp overhead calculation

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documents why it is safe to not follow libsrtp's advice
to ensure additional SRTP_MAX_TRAILER_LEN bytes are available
when calling srtp_protect (and similar srtcp functions).

BUG=None

Change-Id: I504645d21553160f06133fd8bb3ee79e178247da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209064
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33396}
This commit is contained in:
Philipp Hancke
2021-03-01 14:56:22 +01:00
committed by Commit Bot
parent 456a2642d3
commit be66d95ab7
2 changed files with 14 additions and 0 deletions
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8
pc/srtp_session.cc
View File

@ -80,6 +80,10 @@ bool SrtpSession::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
return false;
}
// Note: the need_len differs from the libsrtp recommendatіon to ensure
// SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
// never includes a MKI, therefore the amount of bytes added by the
// srtp_protect call is known in advance and depends on the cipher suite.
int need_len = in_len + rtp_auth_tag_len_; // NOLINT
if (max_len < need_len) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTP packet: The buffer length "
@ -122,6 +126,10 @@ bool SrtpSession::ProtectRtcp(void* p, int in_len, int max_len, int* out_len) {
return false;
}
// Note: the need_len differs from the libsrtp recommendatіon to ensure
// SRTP_MAX_TRAILER_LEN bytes of free space after the data. WebRTC
// never includes a MKI, therefore the amount of bytes added by the
// srtp_protect_rtp call is known in advance and depends on the cipher suite.
int need_len = in_len + sizeof(uint32_t) + rtcp_auth_tag_len_; // NOLINT
if (max_len < need_len) {
RTC_LOG(LS_WARNING) << "Failed to protect SRTCP packet: The buffer length "

6
pc/srtp_session.h
View File

@ -126,8 +126,14 @@ class SrtpSession {
webrtc::SequenceChecker thread_checker_;
srtp_ctx_t_* session_ = nullptr;
// Overhead of the SRTP auth tag for RTP and RTCP in bytes.
// Depends on the cipher suite used and is usually the same with the exception
// of the CS_AES_CM_128_HMAC_SHA1_32 cipher suite. The additional four bytes
// required for RTCP protection are not included.
int rtp_auth_tag_len_ = 0;
int rtcp_auth_tag_len_ = 0;
bool inited_ = false;
static webrtc::GlobalMutex lock_;
int last_send_seq_num_ = -1;