diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 6c7dff322b..2a95a3a816 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -398,10 +398,6 @@ uint8_t* RtpPacket::AllocatePayload(size_t size_bytes) {
 
 uint8_t* RtpPacket::SetPayloadSize(size_t size_bytes) {
   RTC_DCHECK_EQ(padding_size_, 0);
-  if (payload_offset_ + size_bytes > capacity()) {
-    RTC_LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
-    return nullptr;
-  }
   payload_size_ = size_bytes;
   buffer_.SetSize(payload_offset_ + payload_size_);
   return WriteAt(payload_offset_);
diff --git a/modules/rtp_rtcp/source/rtp_packet.h b/modules/rtp_rtcp/source/rtp_packet.h
index 1db4a13017..e91ec6368b 100644
--- a/modules/rtp_rtcp/source/rtp_packet.h
+++ b/modules/rtp_rtcp/source/rtp_packet.h
@@ -153,8 +153,11 @@ class RtpPacket {
   // Returns view of the raw extension or empty view on failure.
   rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
 
-  // Reserve size_bytes for payload. Returns nullptr on failure.
+  // Returns pointer to the payload of size at least `size_bytes`.
+  // Keeps original payload, if any. If `size_bytes` is larger than current
+  // `payload_size()`, remaining bytes are uninitialized.
   uint8_t* SetPayloadSize(size_t size_bytes);
+
   // Same as SetPayloadSize but doesn't guarantee to keep current payload.
   uint8_t* AllocatePayload(size_t size_bytes);
 
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 8c0b2dda0a..967d38a8e0 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -506,15 +506,8 @@ size_t RTPSender::ExpectedPerPacketOverhead() const {
 
 std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
   MutexLock lock(&send_mutex_);
-  // TODO(danilchap): Find better motivator and value for extra capacity.
-  // RtpPacketizer might slightly miscalulate needed size,
-  // SRTP may benefit from extra space in the buffer and do encryption in place
-  // saving reallocation.
-  // While sending slightly oversized packet increase chance of dropped packet,
-  // it is better than crash on drop packet without trying to send it.
-  static constexpr int kExtraCapacity = 16;
-  auto packet = std::make_unique<RtpPacketToSend>(
-      &rtp_header_extension_map_, max_packet_size_ + kExtraCapacity);
+  auto packet = std::make_unique<RtpPacketToSend>(&rtp_header_extension_map_,
+                                                  max_packet_size_);
   packet->SetSsrc(ssrc_);
   packet->SetCsrcs(csrcs_);
 
@@ -721,8 +714,7 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
 
   uint8_t* rtx_payload =
       rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
-  if (rtx_payload == nullptr)
-    return nullptr;
+  RTC_CHECK(rtx_payload);
 
   // Add OSN (original sequence number).
   ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 47b7b7910b..a0f1af5243 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -288,8 +288,7 @@ bool RTPSenderAudio::SendAudio(AudioFrameType frame_type,
   }
 
   uint8_t* payload = packet->AllocatePayload(payload_size);
-  if (!payload)  // Too large payload buffer.
-    return false;
+  RTC_CHECK(payload);
   memcpy(payload, payload_data, payload_size);
 
   {