diff --git a/CHANGELOG.md b/CHANGELOG.md
index afa31f1bc8..dbbf6c418c 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -1,5 +1,9 @@
 # Changelog
 
+### NEXT
+
+- C++: Update libsrtp to v3.0-alpha.
+
 ### 3.13.16
 
 - Node: Add new `worker.on('subprocessclose')` event ([PR #1307](https://github.com/versatica/mediasoup/pull/1307)).
diff --git a/worker/include/RTC/SrtpSession.hpp b/worker/include/RTC/SrtpSession.hpp
index fa14a14760..b50f6d9b87 100644
--- a/worker/include/RTC/SrtpSession.hpp
+++ b/worker/include/RTC/SrtpSession.hpp
@@ -38,13 +38,13 @@ namespace RTC
 		~SrtpSession();
 
 	public:
-		bool EncryptRtp(const uint8_t** data, int* len);
-		bool DecryptSrtp(uint8_t* data, int* len);
-		bool EncryptRtcp(const uint8_t** data, int* len);
-		bool DecryptSrtcp(uint8_t* data, int* len);
+		bool EncryptRtp(const uint8_t** data, size_t* len);
+		bool DecryptSrtp(uint8_t* data, size_t* len);
+		bool EncryptRtcp(const uint8_t** data, size_t* len);
+		bool DecryptSrtcp(uint8_t* data, size_t* len);
 		void RemoveStream(uint32_t ssrc)
 		{
-			srtp_remove_stream(this->session, uint32_t{ htonl(ssrc) });
+			srtp_stream_remove(this->session, uint32_t{ htonl(ssrc) });
 		}
 
 	private:
diff --git a/worker/src/RTC/PipeTransport.cpp b/worker/src/RTC/PipeTransport.cpp
index 8706f5a60b..ff39616228 100644
--- a/worker/src/RTC/PipeTransport.cpp
+++ b/worker/src/RTC/PipeTransport.cpp
@@ -484,9 +484,9 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtp(&data, &len))
 		{
 			if (cb)
 			{
@@ -497,8 +497,6 @@ namespace RTC
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->tuple->Send(data, len, cb);
 
 		// Increase send transmission.
@@ -515,15 +513,13 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->tuple->Send(data, len);
 
 		// Increase send transmission.
@@ -542,15 +538,13 @@ namespace RTC
 		packet->Serialize(RTC::RTCP::Buffer);
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->tuple->Send(data, len);
 
 		// Increase send transmission.
@@ -638,11 +632,9 @@ namespace RTC
 		}
 
 		// Decrypt the SRTP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &intLen))
+		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &len))
 		{
-			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 			if (!packet)
 			{
@@ -663,7 +655,7 @@ namespace RTC
 			return;
 		}
 
-		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 		if (!packet)
 		{
@@ -700,9 +692,7 @@ namespace RTC
 		}
 
 		// Decrypt the SRTCP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &intLen))
+		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &len))
 		{
 			return;
 		}
@@ -715,7 +705,7 @@ namespace RTC
 			return;
 		}
 
-		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, static_cast<size_t>(intLen));
+		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, len);
 
 		if (!packet)
 		{
diff --git a/worker/src/RTC/PlainTransport.cpp b/worker/src/RTC/PlainTransport.cpp
index f985dc8fa8..f6da989dd3 100644
--- a/worker/src/RTC/PlainTransport.cpp
+++ b/worker/src/RTC/PlainTransport.cpp
@@ -764,9 +764,9 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtp(&data, &len))
 		{
 			if (cb)
 			{
@@ -777,8 +777,6 @@ namespace RTC
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->tuple->Send(data, len, cb);
 
 		// Increase send transmission.
@@ -795,15 +793,13 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		if (this->rtcpMux)
 		{
 			this->tuple->Send(data, len);
@@ -829,15 +825,13 @@ namespace RTC
 		packet->Serialize(RTC::RTCP::Buffer);
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (HasSrtp() && !this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		if (this->rtcpMux)
 		{
 			this->tuple->Send(data, len);
@@ -933,11 +927,9 @@ namespace RTC
 		}
 
 		// Decrypt the SRTP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &intLen))
+		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &len))
 		{
-			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 			if (!packet)
 			{
@@ -958,7 +950,7 @@ namespace RTC
 			return;
 		}
 
-		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 		if (!packet)
 		{
@@ -1031,9 +1023,7 @@ namespace RTC
 		}
 
 		// Decrypt the SRTCP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &intLen))
+		if (HasSrtp() && !this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &len))
 		{
 			return;
 		}
@@ -1107,7 +1097,7 @@ namespace RTC
 			return;
 		}
 
-		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, static_cast<size_t>(intLen));
+		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, len);
 
 		if (!packet)
 		{
diff --git a/worker/src/RTC/SrtpSession.cpp b/worker/src/RTC/SrtpSession.cpp
index 5c2c810e0f..20755dc656 100644
--- a/worker/src/RTC/SrtpSession.cpp
+++ b/worker/src/RTC/SrtpSession.cpp
@@ -189,14 +189,14 @@ namespace RTC
 		}
 	}
 
-	bool SrtpSession::EncryptRtp(const uint8_t** data, int* len)
+	bool SrtpSession::EncryptRtp(const uint8_t** data, size_t* len)
 	{
 		MS_TRACE();
 
 		// Ensure that the resulting SRTP packet fits into the encrypt buffer.
-		if (static_cast<size_t>(*len) + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
+		if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
 		{
-			MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%i bytes)", *len);
+			MS_WARN_TAG(srtp, "cannot encrypt RTP packet, size too big (%zu bytes)", *len);
 
 			return false;
 		}
@@ -224,7 +224,7 @@ namespace RTC
 
 		std::memcpy(encryptBuffer, *data, *len);
 
-		const srtp_err_status_t err = srtp_protect(this->session, static_cast<void*>(encryptBuffer), len);
+		const srtp_err_status_t err = srtp_protect(this->session, encryptBuffer, len);
 
 		if (DepLibSRTP::IsError(err))
 		{
@@ -239,11 +239,11 @@ namespace RTC
 		return true;
 	}
 
-	bool SrtpSession::DecryptSrtp(uint8_t* data, int* len)
+	bool SrtpSession::DecryptSrtp(uint8_t* data, size_t* len)
 	{
 		MS_TRACE();
 
-		const srtp_err_status_t err = srtp_unprotect(this->session, static_cast<void*>(data), len);
+		const srtp_err_status_t err = srtp_unprotect(this->session, data, len);
 
 		if (DepLibSRTP::IsError(err))
 		{
@@ -255,22 +255,21 @@ namespace RTC
 		return true;
 	}
 
-	bool SrtpSession::EncryptRtcp(const uint8_t** data, int* len)
+	bool SrtpSession::EncryptRtcp(const uint8_t** data, size_t* len)
 	{
 		MS_TRACE();
 
 		// Ensure that the resulting SRTCP packet fits into the encrypt buffer.
-		if (static_cast<size_t>(*len) + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
+		if (*len + SRTP_MAX_TRAILER_LEN > EncryptBufferSize)
 		{
-			MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%i bytes)", *len);
+			MS_WARN_TAG(srtp, "cannot encrypt RTCP packet, size too big (%zu bytes)", *len);
 
 			return false;
 		}
 
 		std::memcpy(EncryptBuffer, *data, *len);
 
-		const srtp_err_status_t err =
-		  srtp_protect_rtcp(this->session, static_cast<void*>(EncryptBuffer), len);
+		const srtp_err_status_t err = srtp_protect_rtcp(this->session, EncryptBuffer, len);
 
 		if (DepLibSRTP::IsError(err))
 		{
@@ -285,11 +284,11 @@ namespace RTC
 		return true;
 	}
 
-	bool SrtpSession::DecryptSrtcp(uint8_t* data, int* len)
+	bool SrtpSession::DecryptSrtcp(uint8_t* data, size_t* len)
 	{
 		MS_TRACE();
 
-		const srtp_err_status_t err = srtp_unprotect_rtcp(this->session, static_cast<void*>(data), len);
+		const srtp_err_status_t err = srtp_unprotect_rtcp(this->session, data, len);
 
 		if (DepLibSRTP::IsError(err))
 		{
diff --git a/worker/src/RTC/WebRtcTransport.cpp b/worker/src/RTC/WebRtcTransport.cpp
index a795fef798..e699a26ed5 100644
--- a/worker/src/RTC/WebRtcTransport.cpp
+++ b/worker/src/RTC/WebRtcTransport.cpp
@@ -708,9 +708,9 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
-		if (!this->srtpSendSession->EncryptRtp(&data, &intLen))
+		if (!this->srtpSendSession->EncryptRtp(&data, &len))
 		{
 			if (cb)
 			{
@@ -721,8 +721,6 @@ namespace RTC
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->iceServer->GetSelectedTuple()->Send(data, len, cb);
 
 		// Increase send transmission.
@@ -739,7 +737,7 @@ namespace RTC
 		}
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
 		// Ensure there is sending SRTP session.
 		if (!this->srtpSendSession)
@@ -749,13 +747,11 @@ namespace RTC
 			return;
 		}
 
-		if (!this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (!this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->iceServer->GetSelectedTuple()->Send(data, len);
 
 		// Increase send transmission.
@@ -774,7 +770,7 @@ namespace RTC
 		packet->Serialize(RTC::RTCP::Buffer);
 
 		const uint8_t* data = packet->GetData();
-		auto intLen         = static_cast<int>(packet->GetSize());
+		auto len            = packet->GetSize();
 
 		// Ensure there is sending SRTP session.
 		if (!this->srtpSendSession)
@@ -784,13 +780,11 @@ namespace RTC
 			return;
 		}
 
-		if (!this->srtpSendSession->EncryptRtcp(&data, &intLen))
+		if (!this->srtpSendSession->EncryptRtcp(&data, &len))
 		{
 			return;
 		}
 
-		auto len = static_cast<size_t>(intLen);
-
 		this->iceServer->GetSelectedTuple()->Send(data, len);
 
 		// Increase send transmission.
@@ -957,11 +951,9 @@ namespace RTC
 		}
 
 		// Decrypt the SRTP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (!this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &intLen))
+		if (!this->srtpRecvSession->DecryptSrtp(const_cast<uint8_t*>(data), &len))
 		{
-			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+			RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 			if (!packet)
 			{
@@ -982,7 +974,7 @@ namespace RTC
 			return;
 		}
 
-		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, static_cast<size_t>(intLen));
+		RTC::RtpPacket* packet = RTC::RtpPacket::Parse(data, len);
 
 		if (!packet)
 		{
@@ -1028,14 +1020,12 @@ namespace RTC
 		}
 
 		// Decrypt the SRTCP packet.
-		auto intLen = static_cast<int>(len);
-
-		if (!this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &intLen))
+		if (!this->srtpRecvSession->DecryptSrtcp(const_cast<uint8_t*>(data), &len))
 		{
 			return;
 		}
 
-		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, static_cast<size_t>(intLen));
+		RTC::RTCP::Packet* packet = RTC::RTCP::Packet::Parse(data, len);
 
 		if (!packet)
 		{
diff --git a/worker/subprojects/libsrtp2.wrap b/worker/subprojects/libsrtp2.wrap
index aa1b767f2d..d0181794b0 100644
--- a/worker/subprojects/libsrtp2.wrap
+++ b/worker/subprojects/libsrtp2.wrap
@@ -1,8 +1,8 @@
 [wrap-file]
-directory = libsrtp-4c9f0956f2933ac0650208d69c8d897625ba6301
-source_url = https://github.com/versatica/libsrtp/archive/4c9f0956f2933ac0650208d69c8d897625ba6301.zip
-source_filename = libsrtp-4c9f0956f2933ac0650208d69c8d897625ba6301.zip
-source_hash = 4f3af61e26df398569605fc4bcf377587ca2d8bd34b2b4bf9cdb9590cadbd662
+directory = libsrtp-3.0-alpha
+source_url = https://github.com/versatica/libsrtp/archive/v3.0-alpha.zip
+source_filename = libsrtp-3.0-alpha.zip
+source_hash = 946a472b888ca8d51df172def7681f3f9b14768109ffd22af08fddb1be77d2c6
 
 [provide]
 libsrtp2 = libsrtp2_dep