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henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001/*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
pthatcher@webrtc.orge9db7fe2014-12-13 01:56:39 +000011#include "webrtc/p2p/base/pseudotcp.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000012
13#include <stdio.h>
14#include <stdlib.h>
15
andresp@webrtc.orgff689be2015-02-12 11:54:26 +000016#include <algorithm>
kwiberg3ec46792016-04-27 07:22:53 -070017#include <memory>
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000018#include <set>
19
tfarina5237aaf2015-11-10 23:44:30 -080020#include "webrtc/base/arraysize.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000021#include "webrtc/base/basictypes.h"
22#include "webrtc/base/bytebuffer.h"
23#include "webrtc/base/byteorder.h"
24#include "webrtc/base/common.h"
25#include "webrtc/base/logging.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000026#include "webrtc/base/socket.h"
27#include "webrtc/base/stringutils.h"
28#include "webrtc/base/timeutils.h"
29
30// The following logging is for detailed (packet-level) analysis only.
31#define _DBG_NONE 0
32#define _DBG_NORMAL 1
33#define _DBG_VERBOSE 2
34#define _DEBUGMSG _DBG_NONE
35
36namespace cricket {
37
38//////////////////////////////////////////////////////////////////////
39// Network Constants
40//////////////////////////////////////////////////////////////////////
41
42// Standard MTUs
Peter Boström0c4e06b2015-10-07 12:23:21 +020043const uint16_t PACKET_MAXIMUMS[] = {
44 65535, // Theoretical maximum, Hyperchannel
45 32000, // Nothing
46 17914, // 16Mb IBM Token Ring
47 8166, // IEEE 802.4
48 // 4464, // IEEE 802.5 (4Mb max)
49 4352, // FDDI
50 // 2048, // Wideband Network
51 2002, // IEEE 802.5 (4Mb recommended)
52 // 1536, // Expermental Ethernet Networks
53 // 1500, // Ethernet, Point-to-Point (default)
54 1492, // IEEE 802.3
55 1006, // SLIP, ARPANET
56 // 576, // X.25 Networks
57 // 544, // DEC IP Portal
58 // 512, // NETBIOS
59 508, // IEEE 802/Source-Rt Bridge, ARCNET
60 296, // Point-to-Point (low delay)
61 // 68, // Official minimum
62 0, // End of list marker
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000063};
64
Peter Boström0c4e06b2015-10-07 12:23:21 +020065const uint32_t MAX_PACKET = 65535;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000066// Note: we removed lowest level because packet overhead was larger!
Peter Boström0c4e06b2015-10-07 12:23:21 +020067const uint32_t MIN_PACKET = 296;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000068
Peter Boström0c4e06b2015-10-07 12:23:21 +020069const uint32_t IP_HEADER_SIZE = 20; // (+ up to 40 bytes of options?)
70const uint32_t UDP_HEADER_SIZE = 8;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000071// TODO: Make JINGLE_HEADER_SIZE transparent to this code?
Peter Boström0c4e06b2015-10-07 12:23:21 +020072const uint32_t JINGLE_HEADER_SIZE = 64; // when relay framing is in use
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000073
74// Default size for receive and send buffer.
Peter Boström0c4e06b2015-10-07 12:23:21 +020075const uint32_t DEFAULT_RCV_BUF_SIZE = 60 * 1024;
76const uint32_t DEFAULT_SND_BUF_SIZE = 90 * 1024;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000077
78//////////////////////////////////////////////////////////////////////
79// Global Constants and Functions
80//////////////////////////////////////////////////////////////////////
81//
82// 0 1 2 3
83// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
84// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
85// 0 | Conversation Number |
86// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
87// 4 | Sequence Number |
88// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
89// 8 | Acknowledgment Number |
90// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
91// | | |U|A|P|R|S|F| |
92// 12 | Control | |R|C|S|S|Y|I| Window |
93// | | |G|K|H|T|N|N| |
94// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
95// 16 | Timestamp sending |
96// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
97// 20 | Timestamp receiving |
98// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
99// 24 | data |
100// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
101//
102//////////////////////////////////////////////////////////////////////
103
104#define PSEUDO_KEEPALIVE 0
105
Peter Boström0c4e06b2015-10-07 12:23:21 +0200106const uint32_t HEADER_SIZE = 24;
107const uint32_t PACKET_OVERHEAD =
108 HEADER_SIZE + UDP_HEADER_SIZE + IP_HEADER_SIZE + JINGLE_HEADER_SIZE;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000109
Peter Boström0c4e06b2015-10-07 12:23:21 +0200110const uint32_t MIN_RTO =
111 250; // 250 ms (RFC1122, Sec 4.2.3.1 "fractions of a second")
112const uint32_t DEF_RTO = 3000; // 3 seconds (RFC1122, Sec 4.2.3.1)
113const uint32_t MAX_RTO = 60000; // 60 seconds
114const uint32_t DEF_ACK_DELAY = 100; // 100 milliseconds
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000115
Peter Boström0c4e06b2015-10-07 12:23:21 +0200116const uint8_t FLAG_CTL = 0x02;
117const uint8_t FLAG_RST = 0x04;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000118
Peter Boström0c4e06b2015-10-07 12:23:21 +0200119const uint8_t CTL_CONNECT = 0;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000120
121// TCP options.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200122const uint8_t TCP_OPT_EOL = 0; // End of list.
123const uint8_t TCP_OPT_NOOP = 1; // No-op.
124const uint8_t TCP_OPT_MSS = 2; // Maximum segment size.
125const uint8_t TCP_OPT_WND_SCALE = 3; // Window scale factor.
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000126
127const long DEFAULT_TIMEOUT = 4000; // If there are no pending clocks, wake up every 4 seconds
128const long CLOSED_TIMEOUT = 60 * 1000; // If the connection is closed, once per minute
129
130#if PSEUDO_KEEPALIVE
131// !?! Rethink these times
Peter Boström0c4e06b2015-10-07 12:23:21 +0200132const uint32_t IDLE_PING =
133 20 *
134 1000; // 20 seconds (note: WinXP SP2 firewall udp timeout is 90 seconds)
135const uint32_t IDLE_TIMEOUT = 90 * 1000; // 90 seconds;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000136#endif // PSEUDO_KEEPALIVE
137
138//////////////////////////////////////////////////////////////////////
139// Helper Functions
140//////////////////////////////////////////////////////////////////////
141
Peter Boström0c4e06b2015-10-07 12:23:21 +0200142inline void long_to_bytes(uint32_t val, void* buf) {
143 *static_cast<uint32_t*>(buf) = rtc::HostToNetwork32(val);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000144}
145
Peter Boström0c4e06b2015-10-07 12:23:21 +0200146inline void short_to_bytes(uint16_t val, void* buf) {
147 *static_cast<uint16_t*>(buf) = rtc::HostToNetwork16(val);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000148}
149
Peter Boström0c4e06b2015-10-07 12:23:21 +0200150inline uint32_t bytes_to_long(const void* buf) {
151 return rtc::NetworkToHost32(*static_cast<const uint32_t*>(buf));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000152}
153
Peter Boström0c4e06b2015-10-07 12:23:21 +0200154inline uint16_t bytes_to_short(const void* buf) {
155 return rtc::NetworkToHost16(*static_cast<const uint16_t*>(buf));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000156}
157
Peter Boström0c4e06b2015-10-07 12:23:21 +0200158uint32_t bound(uint32_t lower, uint32_t middle, uint32_t upper) {
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000159 return std::min(std::max(lower, middle), upper);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000160}
161
162//////////////////////////////////////////////////////////////////////
163// Debugging Statistics
164//////////////////////////////////////////////////////////////////////
165
166#if 0 // Not used yet
167
168enum Stat {
169 S_SENT_PACKET, // All packet sends
170 S_RESENT_PACKET, // All packet sends that are retransmits
171 S_RECV_PACKET, // All packet receives
172 S_RECV_NEW, // All packet receives that are too new
173 S_RECV_OLD, // All packet receives that are too old
174 S_NUM_STATS
175};
176
177const char* const STAT_NAMES[S_NUM_STATS] = {
178 "snt",
179 "snt-r",
180 "rcv"
181 "rcv-n",
182 "rcv-o"
183};
184
185int g_stats[S_NUM_STATS];
186inline void Incr(Stat s) { ++g_stats[s]; }
187void ReportStats() {
188 char buffer[256];
189 size_t len = 0;
190 for (int i = 0; i < S_NUM_STATS; ++i) {
tfarina5237aaf2015-11-10 23:44:30 -0800191 len += rtc::sprintfn(buffer, arraysize(buffer), "%s%s:%d",
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000192 (i == 0) ? "" : ",", STAT_NAMES[i], g_stats[i]);
193 g_stats[i] = 0;
194 }
195 LOG(LS_INFO) << "Stats[" << buffer << "]";
196}
197
198#endif
199
200//////////////////////////////////////////////////////////////////////
201// PseudoTcp
202//////////////////////////////////////////////////////////////////////
203
Peter Boström0c4e06b2015-10-07 12:23:21 +0200204uint32_t PseudoTcp::Now() {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000205#if 0 // Use this to synchronize timers with logging timestamps (easier debug)
Honghai Zhang82d78622016-05-06 11:29:15 -0700206 return static_cast<uint32_t>(rtc::TimeSince(StartTime()));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000207#else
Honghai Zhang82d78622016-05-06 11:29:15 -0700208 return rtc::Time32();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000209#endif
210}
211
Peter Boström0c4e06b2015-10-07 12:23:21 +0200212PseudoTcp::PseudoTcp(IPseudoTcpNotify* notify, uint32_t conv)
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000213 : m_notify(notify),
214 m_shutdown(SD_NONE),
215 m_error(0),
216 m_rbuf_len(DEFAULT_RCV_BUF_SIZE),
217 m_rbuf(m_rbuf_len),
218 m_sbuf_len(DEFAULT_SND_BUF_SIZE),
219 m_sbuf(m_sbuf_len) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000220 // Sanity check on buffer sizes (needed for OnTcpWriteable notification logic)
221 ASSERT(m_rbuf_len + MIN_PACKET < m_sbuf_len);
222
Peter Boström0c4e06b2015-10-07 12:23:21 +0200223 uint32_t now = Now();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000224
225 m_state = TCP_LISTEN;
226 m_conv = conv;
227 m_rcv_wnd = m_rbuf_len;
228 m_rwnd_scale = m_swnd_scale = 0;
229 m_snd_nxt = 0;
230 m_snd_wnd = 1;
231 m_snd_una = m_rcv_nxt = 0;
232 m_bReadEnable = true;
233 m_bWriteEnable = false;
234 m_t_ack = 0;
235
236 m_msslevel = 0;
237 m_largest = 0;
238 ASSERT(MIN_PACKET > PACKET_OVERHEAD);
239 m_mss = MIN_PACKET - PACKET_OVERHEAD;
240 m_mtu_advise = MAX_PACKET;
241
242 m_rto_base = 0;
243
244 m_cwnd = 2 * m_mss;
245 m_ssthresh = m_rbuf_len;
246 m_lastrecv = m_lastsend = m_lasttraffic = now;
247 m_bOutgoing = false;
248
249 m_dup_acks = 0;
250 m_recover = 0;
251
252 m_ts_recent = m_ts_lastack = 0;
253
254 m_rx_rto = DEF_RTO;
255 m_rx_srtt = m_rx_rttvar = 0;
256
257 m_use_nagling = true;
258 m_ack_delay = DEF_ACK_DELAY;
259 m_support_wnd_scale = true;
260}
261
262PseudoTcp::~PseudoTcp() {
263}
264
265int PseudoTcp::Connect() {
266 if (m_state != TCP_LISTEN) {
267 m_error = EINVAL;
268 return -1;
269 }
270
271 m_state = TCP_SYN_SENT;
272 LOG(LS_INFO) << "State: TCP_SYN_SENT";
273
274 queueConnectMessage();
275 attemptSend();
276
277 return 0;
278}
279
Peter Boström0c4e06b2015-10-07 12:23:21 +0200280void PseudoTcp::NotifyMTU(uint16_t mtu) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000281 m_mtu_advise = mtu;
282 if (m_state == TCP_ESTABLISHED) {
283 adjustMTU();
284 }
285}
286
Peter Boström0c4e06b2015-10-07 12:23:21 +0200287void PseudoTcp::NotifyClock(uint32_t now) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000288 if (m_state == TCP_CLOSED)
289 return;
290
291 // Check if it's time to retransmit a segment
Honghai Zhang82d78622016-05-06 11:29:15 -0700292 if (m_rto_base && (rtc::TimeDiff32(m_rto_base + m_rx_rto, now) <= 0)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000293 if (m_slist.empty()) {
294 ASSERT(false);
295 } else {
296 // Note: (m_slist.front().xmit == 0)) {
297 // retransmit segments
298#if _DEBUGMSG >= _DBG_NORMAL
299 LOG(LS_INFO) << "timeout retransmit (rto: " << m_rx_rto
300 << ") (rto_base: " << m_rto_base
301 << ") (now: " << now
302 << ") (dup_acks: " << static_cast<unsigned>(m_dup_acks)
303 << ")";
304#endif // _DEBUGMSG
305 if (!transmit(m_slist.begin(), now)) {
306 closedown(ECONNABORTED);
307 return;
308 }
309
Peter Boström0c4e06b2015-10-07 12:23:21 +0200310 uint32_t nInFlight = m_snd_nxt - m_snd_una;
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000311 m_ssthresh = std::max(nInFlight / 2, 2 * m_mss);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000312 //LOG(LS_INFO) << "m_ssthresh: " << m_ssthresh << " nInFlight: " << nInFlight << " m_mss: " << m_mss;
313 m_cwnd = m_mss;
314
315 // Back off retransmit timer. Note: the limit is lower when connecting.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200316 uint32_t rto_limit = (m_state < TCP_ESTABLISHED) ? DEF_RTO : MAX_RTO;
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000317 m_rx_rto = std::min(rto_limit, m_rx_rto * 2);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000318 m_rto_base = now;
319 }
320 }
321
322 // Check if it's time to probe closed windows
Honghai Zhang82d78622016-05-06 11:29:15 -0700323 if ((m_snd_wnd == 0) && (rtc::TimeDiff32(m_lastsend + m_rx_rto, now) <= 0)) {
324 if (rtc::TimeDiff32(now, m_lastrecv) >= 15000) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000325 closedown(ECONNABORTED);
326 return;
327 }
328
329 // probe the window
330 packet(m_snd_nxt - 1, 0, 0, 0);
331 m_lastsend = now;
332
333 // back off retransmit timer
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000334 m_rx_rto = std::min(MAX_RTO, m_rx_rto * 2);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000335 }
336
337 // Check if it's time to send delayed acks
Honghai Zhang82d78622016-05-06 11:29:15 -0700338 if (m_t_ack && (rtc::TimeDiff32(m_t_ack + m_ack_delay, now) <= 0)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000339 packet(m_snd_nxt, 0, 0, 0);
340 }
341
342#if PSEUDO_KEEPALIVE
343 // Check for idle timeout
Honghai Zhang82d78622016-05-06 11:29:15 -0700344 if ((m_state == TCP_ESTABLISHED) &&
345 (TimeDiff32(m_lastrecv + IDLE_TIMEOUT, now) <= 0)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000346 closedown(ECONNABORTED);
347 return;
348 }
349
350 // Check for ping timeout (to keep udp mapping open)
Honghai Zhang82d78622016-05-06 11:29:15 -0700351 if ((m_state == TCP_ESTABLISHED) &&
352 (TimeDiff32(m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2 : IDLE_PING),
353 now) <= 0)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000354 packet(m_snd_nxt, 0, 0, 0);
355 }
356#endif // PSEUDO_KEEPALIVE
357}
358
359bool PseudoTcp::NotifyPacket(const char* buffer, size_t len) {
360 if (len > MAX_PACKET) {
361 LOG_F(WARNING) << "packet too large";
362 return false;
363 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200364 return parse(reinterpret_cast<const uint8_t*>(buffer), uint32_t(len));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000365}
366
Peter Boström0c4e06b2015-10-07 12:23:21 +0200367bool PseudoTcp::GetNextClock(uint32_t now, long& timeout) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000368 return clock_check(now, timeout);
369}
370
371void PseudoTcp::GetOption(Option opt, int* value) {
372 if (opt == OPT_NODELAY) {
373 *value = m_use_nagling ? 0 : 1;
374 } else if (opt == OPT_ACKDELAY) {
375 *value = m_ack_delay;
376 } else if (opt == OPT_SNDBUF) {
377 *value = m_sbuf_len;
378 } else if (opt == OPT_RCVBUF) {
379 *value = m_rbuf_len;
380 } else {
381 ASSERT(false);
382 }
383}
384void PseudoTcp::SetOption(Option opt, int value) {
385 if (opt == OPT_NODELAY) {
386 m_use_nagling = value == 0;
387 } else if (opt == OPT_ACKDELAY) {
388 m_ack_delay = value;
389 } else if (opt == OPT_SNDBUF) {
390 ASSERT(m_state == TCP_LISTEN);
391 resizeSendBuffer(value);
392 } else if (opt == OPT_RCVBUF) {
393 ASSERT(m_state == TCP_LISTEN);
394 resizeReceiveBuffer(value);
395 } else {
396 ASSERT(false);
397 }
398}
399
Peter Boström0c4e06b2015-10-07 12:23:21 +0200400uint32_t PseudoTcp::GetCongestionWindow() const {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000401 return m_cwnd;
402}
403
Peter Boström0c4e06b2015-10-07 12:23:21 +0200404uint32_t PseudoTcp::GetBytesInFlight() const {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000405 return m_snd_nxt - m_snd_una;
406}
407
Peter Boström0c4e06b2015-10-07 12:23:21 +0200408uint32_t PseudoTcp::GetBytesBufferedNotSent() const {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000409 size_t buffered_bytes = 0;
410 m_sbuf.GetBuffered(&buffered_bytes);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200411 return static_cast<uint32_t>(m_snd_una + buffered_bytes - m_snd_nxt);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000412}
413
Peter Boström0c4e06b2015-10-07 12:23:21 +0200414uint32_t PseudoTcp::GetRoundTripTimeEstimateMs() const {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000415 return m_rx_srtt;
416}
417
418//
419// IPStream Implementation
420//
421
422int PseudoTcp::Recv(char* buffer, size_t len) {
423 if (m_state != TCP_ESTABLISHED) {
424 m_error = ENOTCONN;
425 return SOCKET_ERROR;
426 }
427
428 size_t read = 0;
429 rtc::StreamResult result = m_rbuf.Read(buffer, len, &read, NULL);
430
431 // If there's no data in |m_rbuf|.
432 if (result == rtc::SR_BLOCK) {
433 m_bReadEnable = true;
434 m_error = EWOULDBLOCK;
435 return SOCKET_ERROR;
436 }
437 ASSERT(result == rtc::SR_SUCCESS);
438
439 size_t available_space = 0;
440 m_rbuf.GetWriteRemaining(&available_space);
441
Peter Boström0c4e06b2015-10-07 12:23:21 +0200442 if (uint32_t(available_space) - m_rcv_wnd >=
443 std::min<uint32_t>(m_rbuf_len / 2, m_mss)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000444 // TODO(jbeda): !?! Not sure about this was closed business
445 bool bWasClosed = (m_rcv_wnd == 0);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200446 m_rcv_wnd = static_cast<uint32_t>(available_space);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000447
448 if (bWasClosed) {
449 attemptSend(sfImmediateAck);
450 }
451 }
452
453 return static_cast<int>(read);
454}
455
456int PseudoTcp::Send(const char* buffer, size_t len) {
457 if (m_state != TCP_ESTABLISHED) {
458 m_error = ENOTCONN;
459 return SOCKET_ERROR;
460 }
461
462 size_t available_space = 0;
463 m_sbuf.GetWriteRemaining(&available_space);
464
465 if (!available_space) {
466 m_bWriteEnable = true;
467 m_error = EWOULDBLOCK;
468 return SOCKET_ERROR;
469 }
470
Peter Boström0c4e06b2015-10-07 12:23:21 +0200471 int written = queue(buffer, uint32_t(len), false);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000472 attemptSend();
473 return written;
474}
475
476void PseudoTcp::Close(bool force) {
477 LOG_F(LS_VERBOSE) << "(" << (force ? "true" : "false") << ")";
478 m_shutdown = force ? SD_FORCEFUL : SD_GRACEFUL;
479}
480
481int PseudoTcp::GetError() {
482 return m_error;
483}
484
485//
486// Internal Implementation
487//
488
Peter Boström0c4e06b2015-10-07 12:23:21 +0200489uint32_t PseudoTcp::queue(const char* data, uint32_t len, bool bCtrl) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000490 size_t available_space = 0;
491 m_sbuf.GetWriteRemaining(&available_space);
492
Peter Boström0c4e06b2015-10-07 12:23:21 +0200493 if (len > static_cast<uint32_t>(available_space)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000494 ASSERT(!bCtrl);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200495 len = static_cast<uint32_t>(available_space);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000496 }
497
498 // We can concatenate data if the last segment is the same type
499 // (control v. regular data), and has not been transmitted yet
500 if (!m_slist.empty() && (m_slist.back().bCtrl == bCtrl) &&
501 (m_slist.back().xmit == 0)) {
502 m_slist.back().len += len;
503 } else {
504 size_t snd_buffered = 0;
505 m_sbuf.GetBuffered(&snd_buffered);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200506 SSegment sseg(static_cast<uint32_t>(m_snd_una + snd_buffered), len, bCtrl);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000507 m_slist.push_back(sseg);
508 }
509
510 size_t written = 0;
511 m_sbuf.Write(data, len, &written, NULL);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200512 return static_cast<uint32_t>(written);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000513}
514
Peter Boström0c4e06b2015-10-07 12:23:21 +0200515IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32_t seq,
516 uint8_t flags,
517 uint32_t offset,
518 uint32_t len) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000519 ASSERT(HEADER_SIZE + len <= MAX_PACKET);
520
Peter Boström0c4e06b2015-10-07 12:23:21 +0200521 uint32_t now = Now();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000522
kwiberg3ec46792016-04-27 07:22:53 -0700523 std::unique_ptr<uint8_t[]> buffer(new uint8_t[MAX_PACKET]);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000524 long_to_bytes(m_conv, buffer.get());
525 long_to_bytes(seq, buffer.get() + 4);
526 long_to_bytes(m_rcv_nxt, buffer.get() + 8);
527 buffer[12] = 0;
528 buffer[13] = flags;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200529 short_to_bytes(static_cast<uint16_t>(m_rcv_wnd >> m_rwnd_scale),
530 buffer.get() + 14);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000531
532 // Timestamp computations
533 long_to_bytes(now, buffer.get() + 16);
534 long_to_bytes(m_ts_recent, buffer.get() + 20);
535 m_ts_lastack = m_rcv_nxt;
536
537 if (len) {
538 size_t bytes_read = 0;
539 rtc::StreamResult result = m_sbuf.ReadOffset(
540 buffer.get() + HEADER_SIZE, len, offset, &bytes_read);
541 RTC_UNUSED(result);
542 ASSERT(result == rtc::SR_SUCCESS);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200543 ASSERT(static_cast<uint32_t>(bytes_read) == len);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000544 }
545
546#if _DEBUGMSG >= _DBG_VERBOSE
547 LOG(LS_INFO) << "<-- <CONV=" << m_conv
548 << "><FLG=" << static_cast<unsigned>(flags)
549 << "><SEQ=" << seq << ":" << seq + len
550 << "><ACK=" << m_rcv_nxt
551 << "><WND=" << m_rcv_wnd
552 << "><TS=" << (now % 10000)
553 << "><TSR=" << (m_ts_recent % 10000)
554 << "><LEN=" << len << ">";
555#endif // _DEBUGMSG
556
557 IPseudoTcpNotify::WriteResult wres = m_notify->TcpWritePacket(
558 this, reinterpret_cast<char *>(buffer.get()), len + HEADER_SIZE);
559 // Note: When len is 0, this is an ACK packet. We don't read the return value for those,
560 // and thus we won't retry. So go ahead and treat the packet as a success (basically simulate
561 // as if it were dropped), which will prevent our timers from being messed up.
562 if ((wres != IPseudoTcpNotify::WR_SUCCESS) && (0 != len))
563 return wres;
564
565 m_t_ack = 0;
566 if (len > 0) {
567 m_lastsend = now;
568 }
569 m_lasttraffic = now;
570 m_bOutgoing = true;
571
572 return IPseudoTcpNotify::WR_SUCCESS;
573}
574
Peter Boström0c4e06b2015-10-07 12:23:21 +0200575bool PseudoTcp::parse(const uint8_t* buffer, uint32_t size) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000576 if (size < 12)
577 return false;
578
579 Segment seg;
580 seg.conv = bytes_to_long(buffer);
581 seg.seq = bytes_to_long(buffer + 4);
582 seg.ack = bytes_to_long(buffer + 8);
583 seg.flags = buffer[13];
584 seg.wnd = bytes_to_short(buffer + 14);
585
586 seg.tsval = bytes_to_long(buffer + 16);
587 seg.tsecr = bytes_to_long(buffer + 20);
588
589 seg.data = reinterpret_cast<const char *>(buffer) + HEADER_SIZE;
590 seg.len = size - HEADER_SIZE;
591
592#if _DEBUGMSG >= _DBG_VERBOSE
593 LOG(LS_INFO) << "--> <CONV=" << seg.conv
594 << "><FLG=" << static_cast<unsigned>(seg.flags)
595 << "><SEQ=" << seg.seq << ":" << seg.seq + seg.len
596 << "><ACK=" << seg.ack
597 << "><WND=" << seg.wnd
598 << "><TS=" << (seg.tsval % 10000)
599 << "><TSR=" << (seg.tsecr % 10000)
600 << "><LEN=" << seg.len << ">";
601#endif // _DEBUGMSG
602
603 return process(seg);
604}
605
Peter Boström0c4e06b2015-10-07 12:23:21 +0200606bool PseudoTcp::clock_check(uint32_t now, long& nTimeout) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000607 if (m_shutdown == SD_FORCEFUL)
608 return false;
609
610 size_t snd_buffered = 0;
611 m_sbuf.GetBuffered(&snd_buffered);
612 if ((m_shutdown == SD_GRACEFUL)
613 && ((m_state != TCP_ESTABLISHED)
614 || ((snd_buffered == 0) && (m_t_ack == 0)))) {
615 return false;
616 }
617
618 if (m_state == TCP_CLOSED) {
619 nTimeout = CLOSED_TIMEOUT;
620 return true;
621 }
622
623 nTimeout = DEFAULT_TIMEOUT;
624
625 if (m_t_ack) {
Honghai Zhang82d78622016-05-06 11:29:15 -0700626 nTimeout = std::min<int32_t>(nTimeout,
627 rtc::TimeDiff32(m_t_ack + m_ack_delay, now));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000628 }
629 if (m_rto_base) {
Honghai Zhang82d78622016-05-06 11:29:15 -0700630 nTimeout = std::min<int32_t>(nTimeout,
631 rtc::TimeDiff32(m_rto_base + m_rx_rto, now));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000632 }
633 if (m_snd_wnd == 0) {
Honghai Zhang82d78622016-05-06 11:29:15 -0700634 nTimeout = std::min<int32_t>(nTimeout,
635 rtc::TimeDiff32(m_lastsend + m_rx_rto, now));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000636 }
637#if PSEUDO_KEEPALIVE
638 if (m_state == TCP_ESTABLISHED) {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200639 nTimeout = std::min<int32_t>(
Honghai Zhang82d78622016-05-06 11:29:15 -0700640 nTimeout,
641 rtc::TimeDiff32(
642 m_lasttraffic + (m_bOutgoing ? IDLE_PING * 3 / 2 : IDLE_PING),
643 now));
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000644 }
645#endif // PSEUDO_KEEPALIVE
646 return true;
647}
648
649bool PseudoTcp::process(Segment& seg) {
650 // If this is the wrong conversation, send a reset!?! (with the correct conversation?)
651 if (seg.conv != m_conv) {
652 //if ((seg.flags & FLAG_RST) == 0) {
653 // packet(tcb, seg.ack, 0, FLAG_RST, 0, 0);
654 //}
655 LOG_F(LS_ERROR) << "wrong conversation";
656 return false;
657 }
658
Peter Boström0c4e06b2015-10-07 12:23:21 +0200659 uint32_t now = Now();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000660 m_lasttraffic = m_lastrecv = now;
661 m_bOutgoing = false;
662
663 if (m_state == TCP_CLOSED) {
664 // !?! send reset?
665 LOG_F(LS_ERROR) << "closed";
666 return false;
667 }
668
669 // Check if this is a reset segment
670 if (seg.flags & FLAG_RST) {
671 closedown(ECONNRESET);
672 return false;
673 }
674
675 // Check for control data
676 bool bConnect = false;
677 if (seg.flags & FLAG_CTL) {
678 if (seg.len == 0) {
679 LOG_F(LS_ERROR) << "Missing control code";
680 return false;
681 } else if (seg.data[0] == CTL_CONNECT) {
682 bConnect = true;
683
684 // TCP options are in the remainder of the payload after CTL_CONNECT.
685 parseOptions(&seg.data[1], seg.len - 1);
686
687 if (m_state == TCP_LISTEN) {
688 m_state = TCP_SYN_RECEIVED;
689 LOG(LS_INFO) << "State: TCP_SYN_RECEIVED";
690 //m_notify->associate(addr);
691 queueConnectMessage();
692 } else if (m_state == TCP_SYN_SENT) {
693 m_state = TCP_ESTABLISHED;
694 LOG(LS_INFO) << "State: TCP_ESTABLISHED";
695 adjustMTU();
696 if (m_notify) {
697 m_notify->OnTcpOpen(this);
698 }
699 //notify(evOpen);
700 }
701 } else {
702 LOG_F(LS_WARNING) << "Unknown control code: " << seg.data[0];
703 return false;
704 }
705 }
706
707 // Update timestamp
708 if ((seg.seq <= m_ts_lastack) && (m_ts_lastack < seg.seq + seg.len)) {
709 m_ts_recent = seg.tsval;
710 }
711
712 // Check if this is a valuable ack
713 if ((seg.ack > m_snd_una) && (seg.ack <= m_snd_nxt)) {
714 // Calculate round-trip time
715 if (seg.tsecr) {
Honghai Zhang82d78622016-05-06 11:29:15 -0700716 int32_t rtt = rtc::TimeDiff32(now, seg.tsecr);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000717 if (rtt >= 0) {
718 if (m_rx_srtt == 0) {
719 m_rx_srtt = rtt;
720 m_rx_rttvar = rtt / 2;
721 } else {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200722 uint32_t unsigned_rtt = static_cast<uint32_t>(rtt);
723 uint32_t abs_err = unsigned_rtt > m_rx_srtt
724 ? unsigned_rtt - m_rx_srtt
725 : m_rx_srtt - unsigned_rtt;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000726 m_rx_rttvar = (3 * m_rx_rttvar + abs_err) / 4;
727 m_rx_srtt = (7 * m_rx_srtt + rtt) / 8;
728 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200729 m_rx_rto =
730 bound(MIN_RTO, m_rx_srtt + std::max<uint32_t>(1, 4 * m_rx_rttvar),
731 MAX_RTO);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000732#if _DEBUGMSG >= _DBG_VERBOSE
733 LOG(LS_INFO) << "rtt: " << rtt
734 << " srtt: " << m_rx_srtt
735 << " rto: " << m_rx_rto;
736#endif // _DEBUGMSG
737 } else {
738 ASSERT(false);
739 }
740 }
741
Peter Boström0c4e06b2015-10-07 12:23:21 +0200742 m_snd_wnd = static_cast<uint32_t>(seg.wnd) << m_swnd_scale;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000743
Peter Boström0c4e06b2015-10-07 12:23:21 +0200744 uint32_t nAcked = seg.ack - m_snd_una;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000745 m_snd_una = seg.ack;
746
747 m_rto_base = (m_snd_una == m_snd_nxt) ? 0 : now;
748
749 m_sbuf.ConsumeReadData(nAcked);
750
Peter Boström0c4e06b2015-10-07 12:23:21 +0200751 for (uint32_t nFree = nAcked; nFree > 0;) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000752 ASSERT(!m_slist.empty());
753 if (nFree < m_slist.front().len) {
754 m_slist.front().len -= nFree;
755 nFree = 0;
756 } else {
757 if (m_slist.front().len > m_largest) {
758 m_largest = m_slist.front().len;
759 }
760 nFree -= m_slist.front().len;
761 m_slist.pop_front();
762 }
763 }
764
765 if (m_dup_acks >= 3) {
766 if (m_snd_una >= m_recover) { // NewReno
Peter Boström0c4e06b2015-10-07 12:23:21 +0200767 uint32_t nInFlight = m_snd_nxt - m_snd_una;
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000768 m_cwnd = std::min(m_ssthresh, nInFlight + m_mss); // (Fast Retransmit)
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000769#if _DEBUGMSG >= _DBG_NORMAL
770 LOG(LS_INFO) << "exit recovery";
771#endif // _DEBUGMSG
772 m_dup_acks = 0;
773 } else {
774#if _DEBUGMSG >= _DBG_NORMAL
775 LOG(LS_INFO) << "recovery retransmit";
776#endif // _DEBUGMSG
777 if (!transmit(m_slist.begin(), now)) {
778 closedown(ECONNABORTED);
779 return false;
780 }
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000781 m_cwnd += m_mss - std::min(nAcked, m_cwnd);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000782 }
783 } else {
784 m_dup_acks = 0;
785 // Slow start, congestion avoidance
786 if (m_cwnd < m_ssthresh) {
787 m_cwnd += m_mss;
788 } else {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200789 m_cwnd += std::max<uint32_t>(1, m_mss * m_mss / m_cwnd);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000790 }
791 }
792 } else if (seg.ack == m_snd_una) {
793 // !?! Note, tcp says don't do this... but otherwise how does a closed window become open?
Peter Boström0c4e06b2015-10-07 12:23:21 +0200794 m_snd_wnd = static_cast<uint32_t>(seg.wnd) << m_swnd_scale;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000795
796 // Check duplicate acks
797 if (seg.len > 0) {
798 // it's a dup ack, but with a data payload, so don't modify m_dup_acks
799 } else if (m_snd_una != m_snd_nxt) {
800 m_dup_acks += 1;
801 if (m_dup_acks == 3) { // (Fast Retransmit)
802#if _DEBUGMSG >= _DBG_NORMAL
803 LOG(LS_INFO) << "enter recovery";
804 LOG(LS_INFO) << "recovery retransmit";
805#endif // _DEBUGMSG
806 if (!transmit(m_slist.begin(), now)) {
807 closedown(ECONNABORTED);
808 return false;
809 }
810 m_recover = m_snd_nxt;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200811 uint32_t nInFlight = m_snd_nxt - m_snd_una;
andresp@webrtc.orgff689be2015-02-12 11:54:26 +0000812 m_ssthresh = std::max(nInFlight / 2, 2 * m_mss);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000813 //LOG(LS_INFO) << "m_ssthresh: " << m_ssthresh << " nInFlight: " << nInFlight << " m_mss: " << m_mss;
814 m_cwnd = m_ssthresh + 3 * m_mss;
815 } else if (m_dup_acks > 3) {
816 m_cwnd += m_mss;
817 }
818 } else {
819 m_dup_acks = 0;
820 }
821 }
822
823 // !?! A bit hacky
824 if ((m_state == TCP_SYN_RECEIVED) && !bConnect) {
825 m_state = TCP_ESTABLISHED;
826 LOG(LS_INFO) << "State: TCP_ESTABLISHED";
827 adjustMTU();
828 if (m_notify) {
829 m_notify->OnTcpOpen(this);
830 }
831 //notify(evOpen);
832 }
833
834 // If we make room in the send queue, notify the user
835 // The goal it to make sure we always have at least enough data to fill the
836 // window. We'd like to notify the app when we are halfway to that point.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200837 const uint32_t kIdealRefillSize = (m_sbuf_len + m_rbuf_len) / 2;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000838 size_t snd_buffered = 0;
839 m_sbuf.GetBuffered(&snd_buffered);
Peter Boström0c4e06b2015-10-07 12:23:21 +0200840 if (m_bWriteEnable &&
841 static_cast<uint32_t>(snd_buffered) < kIdealRefillSize) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000842 m_bWriteEnable = false;
843 if (m_notify) {
844 m_notify->OnTcpWriteable(this);
845 }
846 //notify(evWrite);
847 }
848
849 // Conditions were acks must be sent:
850 // 1) Segment is too old (they missed an ACK) (immediately)
851 // 2) Segment is too new (we missed a segment) (immediately)
852 // 3) Segment has data (so we need to ACK!) (delayed)
853 // ... so the only time we don't need to ACK, is an empty segment that points to rcv_nxt!
854
855 SendFlags sflags = sfNone;
856 if (seg.seq != m_rcv_nxt) {
857 sflags = sfImmediateAck; // (Fast Recovery)
858 } else if (seg.len != 0) {
859 if (m_ack_delay == 0) {
860 sflags = sfImmediateAck;
861 } else {
862 sflags = sfDelayedAck;
863 }
864 }
865#if _DEBUGMSG >= _DBG_NORMAL
866 if (sflags == sfImmediateAck) {
867 if (seg.seq > m_rcv_nxt) {
868 LOG_F(LS_INFO) << "too new";
869 } else if (seg.seq + seg.len <= m_rcv_nxt) {
870 LOG_F(LS_INFO) << "too old";
871 }
872 }
873#endif // _DEBUGMSG
874
875 // Adjust the incoming segment to fit our receive buffer
876 if (seg.seq < m_rcv_nxt) {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200877 uint32_t nAdjust = m_rcv_nxt - seg.seq;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000878 if (nAdjust < seg.len) {
879 seg.seq += nAdjust;
880 seg.data += nAdjust;
881 seg.len -= nAdjust;
882 } else {
883 seg.len = 0;
884 }
885 }
886
887 size_t available_space = 0;
888 m_rbuf.GetWriteRemaining(&available_space);
889
Peter Boström0c4e06b2015-10-07 12:23:21 +0200890 if ((seg.seq + seg.len - m_rcv_nxt) >
891 static_cast<uint32_t>(available_space)) {
892 uint32_t nAdjust =
893 seg.seq + seg.len - m_rcv_nxt - static_cast<uint32_t>(available_space);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000894 if (nAdjust < seg.len) {
895 seg.len -= nAdjust;
896 } else {
897 seg.len = 0;
898 }
899 }
900
901 bool bIgnoreData = (seg.flags & FLAG_CTL) || (m_shutdown != SD_NONE);
902 bool bNewData = false;
903
904 if (seg.len > 0) {
905 if (bIgnoreData) {
906 if (seg.seq == m_rcv_nxt) {
907 m_rcv_nxt += seg.len;
908 }
909 } else {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200910 uint32_t nOffset = seg.seq - m_rcv_nxt;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000911
912 rtc::StreamResult result = m_rbuf.WriteOffset(seg.data, seg.len,
913 nOffset, NULL);
914 ASSERT(result == rtc::SR_SUCCESS);
915 RTC_UNUSED(result);
916
917 if (seg.seq == m_rcv_nxt) {
918 m_rbuf.ConsumeWriteBuffer(seg.len);
919 m_rcv_nxt += seg.len;
920 m_rcv_wnd -= seg.len;
921 bNewData = true;
922
923 RList::iterator it = m_rlist.begin();
924 while ((it != m_rlist.end()) && (it->seq <= m_rcv_nxt)) {
925 if (it->seq + it->len > m_rcv_nxt) {
926 sflags = sfImmediateAck; // (Fast Recovery)
Peter Boström0c4e06b2015-10-07 12:23:21 +0200927 uint32_t nAdjust = (it->seq + it->len) - m_rcv_nxt;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000928#if _DEBUGMSG >= _DBG_NORMAL
929 LOG(LS_INFO) << "Recovered " << nAdjust << " bytes (" << m_rcv_nxt << " -> " << m_rcv_nxt + nAdjust << ")";
930#endif // _DEBUGMSG
931 m_rbuf.ConsumeWriteBuffer(nAdjust);
932 m_rcv_nxt += nAdjust;
933 m_rcv_wnd -= nAdjust;
934 }
935 it = m_rlist.erase(it);
936 }
937 } else {
938#if _DEBUGMSG >= _DBG_NORMAL
939 LOG(LS_INFO) << "Saving " << seg.len << " bytes (" << seg.seq << " -> " << seg.seq + seg.len << ")";
940#endif // _DEBUGMSG
941 RSegment rseg;
942 rseg.seq = seg.seq;
943 rseg.len = seg.len;
944 RList::iterator it = m_rlist.begin();
945 while ((it != m_rlist.end()) && (it->seq < rseg.seq)) {
946 ++it;
947 }
948 m_rlist.insert(it, rseg);
949 }
950 }
951 }
952
953 attemptSend(sflags);
954
955 // If we have new data, notify the user
956 if (bNewData && m_bReadEnable) {
957 m_bReadEnable = false;
958 if (m_notify) {
959 m_notify->OnTcpReadable(this);
960 }
961 //notify(evRead);
962 }
963
964 return true;
965}
966
Peter Boström0c4e06b2015-10-07 12:23:21 +0200967bool PseudoTcp::transmit(const SList::iterator& seg, uint32_t now) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000968 if (seg->xmit >= ((m_state == TCP_ESTABLISHED) ? 15 : 30)) {
969 LOG_F(LS_VERBOSE) << "too many retransmits";
970 return false;
971 }
972
Peter Boström0c4e06b2015-10-07 12:23:21 +0200973 uint32_t nTransmit = std::min(seg->len, m_mss);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000974
975 while (true) {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200976 uint32_t seq = seg->seq;
977 uint8_t flags = (seg->bCtrl ? FLAG_CTL : 0);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +0000978 IPseudoTcpNotify::WriteResult wres = packet(seq,
979 flags,
980 seg->seq - m_snd_una,
981 nTransmit);
982
983 if (wres == IPseudoTcpNotify::WR_SUCCESS)
984 break;
985
986 if (wres == IPseudoTcpNotify::WR_FAIL) {
987 LOG_F(LS_VERBOSE) << "packet failed";
988 return false;
989 }
990
991 ASSERT(wres == IPseudoTcpNotify::WR_TOO_LARGE);
992
993 while (true) {
994 if (PACKET_MAXIMUMS[m_msslevel + 1] == 0) {
995 LOG_F(LS_VERBOSE) << "MTU too small";
996 return false;
997 }
998 // !?! We need to break up all outstanding and pending packets and then retransmit!?!
999
1000 m_mss = PACKET_MAXIMUMS[++m_msslevel] - PACKET_OVERHEAD;
1001 m_cwnd = 2 * m_mss; // I added this... haven't researched actual formula
1002 if (m_mss < nTransmit) {
1003 nTransmit = m_mss;
1004 break;
1005 }
1006 }
1007#if _DEBUGMSG >= _DBG_NORMAL
1008 LOG(LS_INFO) << "Adjusting mss to " << m_mss << " bytes";
1009#endif // _DEBUGMSG
1010 }
1011
1012 if (nTransmit < seg->len) {
1013 LOG_F(LS_VERBOSE) << "mss reduced to " << m_mss;
1014
1015 SSegment subseg(seg->seq + nTransmit, seg->len - nTransmit, seg->bCtrl);
1016 //subseg.tstamp = seg->tstamp;
1017 subseg.xmit = seg->xmit;
1018 seg->len = nTransmit;
1019
1020 SList::iterator next = seg;
1021 m_slist.insert(++next, subseg);
1022 }
1023
1024 if (seg->xmit == 0) {
1025 m_snd_nxt += seg->len;
1026 }
1027 seg->xmit += 1;
1028 //seg->tstamp = now;
1029 if (m_rto_base == 0) {
1030 m_rto_base = now;
1031 }
1032
1033 return true;
1034}
1035
1036void PseudoTcp::attemptSend(SendFlags sflags) {
Peter Boström0c4e06b2015-10-07 12:23:21 +02001037 uint32_t now = Now();
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001038
Honghai Zhang82d78622016-05-06 11:29:15 -07001039 if (rtc::TimeDiff32(now, m_lastsend) > static_cast<long>(m_rx_rto)) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001040 m_cwnd = m_mss;
1041 }
1042
1043#if _DEBUGMSG
1044 bool bFirst = true;
1045 RTC_UNUSED(bFirst);
1046#endif // _DEBUGMSG
1047
1048 while (true) {
Peter Boström0c4e06b2015-10-07 12:23:21 +02001049 uint32_t cwnd = m_cwnd;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001050 if ((m_dup_acks == 1) || (m_dup_acks == 2)) { // Limited Transmit
1051 cwnd += m_dup_acks * m_mss;
1052 }
Peter Boström0c4e06b2015-10-07 12:23:21 +02001053 uint32_t nWindow = std::min(m_snd_wnd, cwnd);
1054 uint32_t nInFlight = m_snd_nxt - m_snd_una;
1055 uint32_t nUseable = (nInFlight < nWindow) ? (nWindow - nInFlight) : 0;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001056
1057 size_t snd_buffered = 0;
1058 m_sbuf.GetBuffered(&snd_buffered);
Peter Boström0c4e06b2015-10-07 12:23:21 +02001059 uint32_t nAvailable =
1060 std::min(static_cast<uint32_t>(snd_buffered) - nInFlight, m_mss);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001061
1062 if (nAvailable > nUseable) {
1063 if (nUseable * 4 < nWindow) {
1064 // RFC 813 - avoid SWS
1065 nAvailable = 0;
1066 } else {
1067 nAvailable = nUseable;
1068 }
1069 }
1070
1071#if _DEBUGMSG >= _DBG_VERBOSE
1072 if (bFirst) {
1073 size_t available_space = 0;
1074 m_sbuf.GetWriteRemaining(&available_space);
1075
1076 bFirst = false;
1077 LOG(LS_INFO) << "[cwnd: " << m_cwnd
1078 << " nWindow: " << nWindow
1079 << " nInFlight: " << nInFlight
1080 << " nAvailable: " << nAvailable
1081 << " nQueued: " << snd_buffered
1082 << " nEmpty: " << available_space
1083 << " ssthresh: " << m_ssthresh << "]";
1084 }
1085#endif // _DEBUGMSG
1086
1087 if (nAvailable == 0) {
1088 if (sflags == sfNone)
1089 return;
1090
1091 // If this is an immediate ack, or the second delayed ack
1092 if ((sflags == sfImmediateAck) || m_t_ack) {
1093 packet(m_snd_nxt, 0, 0, 0);
1094 } else {
1095 m_t_ack = Now();
1096 }
1097 return;
1098 }
1099
1100 // Nagle's algorithm.
1101 // If there is data already in-flight, and we haven't a full segment of
1102 // data ready to send then hold off until we get more to send, or the
1103 // in-flight data is acknowledged.
1104 if (m_use_nagling && (m_snd_nxt > m_snd_una) && (nAvailable < m_mss)) {
1105 return;
1106 }
1107
1108 // Find the next segment to transmit
1109 SList::iterator it = m_slist.begin();
1110 while (it->xmit > 0) {
1111 ++it;
1112 ASSERT(it != m_slist.end());
1113 }
1114 SList::iterator seg = it;
1115
1116 // If the segment is too large, break it into two
1117 if (seg->len > nAvailable) {
1118 SSegment subseg(seg->seq + nAvailable, seg->len - nAvailable, seg->bCtrl);
1119 seg->len = nAvailable;
1120 m_slist.insert(++it, subseg);
1121 }
1122
1123 if (!transmit(seg, now)) {
1124 LOG_F(LS_VERBOSE) << "transmit failed";
1125 // TODO: consider closing socket
1126 return;
1127 }
1128
1129 sflags = sfNone;
1130 }
1131}
1132
Peter Boström0c4e06b2015-10-07 12:23:21 +02001133void PseudoTcp::closedown(uint32_t err) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001134 LOG(LS_INFO) << "State: TCP_CLOSED";
1135 m_state = TCP_CLOSED;
1136 if (m_notify) {
1137 m_notify->OnTcpClosed(this, err);
1138 }
1139 //notify(evClose, err);
1140}
1141
1142void
1143PseudoTcp::adjustMTU() {
1144 // Determine our current mss level, so that we can adjust appropriately later
1145 for (m_msslevel = 0; PACKET_MAXIMUMS[m_msslevel + 1] > 0; ++m_msslevel) {
Peter Boström0c4e06b2015-10-07 12:23:21 +02001146 if (static_cast<uint16_t>(PACKET_MAXIMUMS[m_msslevel]) <= m_mtu_advise) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001147 break;
1148 }
1149 }
1150 m_mss = m_mtu_advise - PACKET_OVERHEAD;
1151 // !?! Should we reset m_largest here?
1152#if _DEBUGMSG >= _DBG_NORMAL
1153 LOG(LS_INFO) << "Adjusting mss to " << m_mss << " bytes";
1154#endif // _DEBUGMSG
1155 // Enforce minimums on ssthresh and cwnd
andresp@webrtc.orgff689be2015-02-12 11:54:26 +00001156 m_ssthresh = std::max(m_ssthresh, 2 * m_mss);
1157 m_cwnd = std::max(m_cwnd, m_mss);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001158}
1159
1160bool
1161PseudoTcp::isReceiveBufferFull() const {
1162 size_t available_space = 0;
1163 m_rbuf.GetWriteRemaining(&available_space);
1164 return !available_space;
1165}
1166
1167void
1168PseudoTcp::disableWindowScale() {
1169 m_support_wnd_scale = false;
1170}
1171
1172void
1173PseudoTcp::queueConnectMessage() {
jbauchf1f87202016-03-30 06:43:37 -07001174 rtc::ByteBufferWriter buf(rtc::ByteBuffer::ORDER_NETWORK);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001175
1176 buf.WriteUInt8(CTL_CONNECT);
1177 if (m_support_wnd_scale) {
1178 buf.WriteUInt8(TCP_OPT_WND_SCALE);
1179 buf.WriteUInt8(1);
1180 buf.WriteUInt8(m_rwnd_scale);
1181 }
Peter Boström0c4e06b2015-10-07 12:23:21 +02001182 m_snd_wnd = static_cast<uint32_t>(buf.Length());
1183 queue(buf.Data(), static_cast<uint32_t>(buf.Length()), true);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001184}
1185
Peter Boström0c4e06b2015-10-07 12:23:21 +02001186void PseudoTcp::parseOptions(const char* data, uint32_t len) {
1187 std::set<uint8_t> options_specified;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001188
1189 // See http://www.freesoft.org/CIE/Course/Section4/8.htm for
1190 // parsing the options list.
jbauchf1f87202016-03-30 06:43:37 -07001191 rtc::ByteBufferReader buf(data, len);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001192 while (buf.Length()) {
Peter Boström0c4e06b2015-10-07 12:23:21 +02001193 uint8_t kind = TCP_OPT_EOL;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001194 buf.ReadUInt8(&kind);
1195
1196 if (kind == TCP_OPT_EOL) {
1197 // End of option list.
1198 break;
1199 } else if (kind == TCP_OPT_NOOP) {
1200 // No op.
1201 continue;
1202 }
1203
1204 // Length of this option.
1205 ASSERT(len != 0);
1206 RTC_UNUSED(len);
Peter Boström0c4e06b2015-10-07 12:23:21 +02001207 uint8_t opt_len = 0;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001208 buf.ReadUInt8(&opt_len);
1209
1210 // Content of this option.
1211 if (opt_len <= buf.Length()) {
1212 applyOption(kind, buf.Data(), opt_len);
1213 buf.Consume(opt_len);
1214 } else {
1215 LOG(LS_ERROR) << "Invalid option length received.";
1216 return;
1217 }
1218 options_specified.insert(kind);
1219 }
1220
1221 if (options_specified.find(TCP_OPT_WND_SCALE) == options_specified.end()) {
1222 LOG(LS_WARNING) << "Peer doesn't support window scaling";
1223
1224 if (m_rwnd_scale > 0) {
1225 // Peer doesn't support TCP options and window scaling.
1226 // Revert receive buffer size to default value.
1227 resizeReceiveBuffer(DEFAULT_RCV_BUF_SIZE);
1228 m_swnd_scale = 0;
1229 }
1230 }
1231}
1232
Peter Boström0c4e06b2015-10-07 12:23:21 +02001233void PseudoTcp::applyOption(char kind, const char* data, uint32_t len) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001234 if (kind == TCP_OPT_MSS) {
1235 LOG(LS_WARNING) << "Peer specified MSS option which is not supported.";
1236 // TODO: Implement.
1237 } else if (kind == TCP_OPT_WND_SCALE) {
1238 // Window scale factor.
1239 // http://www.ietf.org/rfc/rfc1323.txt
1240 if (len != 1) {
1241 LOG_F(WARNING) << "Invalid window scale option received.";
1242 return;
1243 }
1244 applyWindowScaleOption(data[0]);
1245 }
1246}
1247
Peter Boström0c4e06b2015-10-07 12:23:21 +02001248void PseudoTcp::applyWindowScaleOption(uint8_t scale_factor) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001249 m_swnd_scale = scale_factor;
1250}
1251
Peter Boström0c4e06b2015-10-07 12:23:21 +02001252void PseudoTcp::resizeSendBuffer(uint32_t new_size) {
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001253 m_sbuf_len = new_size;
1254 m_sbuf.SetCapacity(new_size);
1255}
1256
Peter Boström0c4e06b2015-10-07 12:23:21 +02001257void PseudoTcp::resizeReceiveBuffer(uint32_t new_size) {
1258 uint8_t scale_factor = 0;
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001259
1260 // Determine the scale factor such that the scaled window size can fit
1261 // in a 16-bit unsigned integer.
1262 while (new_size > 0xFFFF) {
1263 ++scale_factor;
1264 new_size >>= 1;
1265 }
1266
1267 // Determine the proper size of the buffer.
1268 new_size <<= scale_factor;
1269 bool result = m_rbuf.SetCapacity(new_size);
1270
1271 // Make sure the new buffer is large enough to contain data in the old
1272 // buffer. This should always be true because this method is called either
1273 // before connection is established or when peers are exchanging connect
1274 // messages.
1275 ASSERT(result);
1276 RTC_UNUSED(result);
1277 m_rbuf_len = new_size;
1278 m_rwnd_scale = scale_factor;
1279 m_ssthresh = new_size;
1280
1281 size_t available_space = 0;
1282 m_rbuf.GetWriteRemaining(&available_space);
Peter Boström0c4e06b2015-10-07 12:23:21 +02001283 m_rcv_wnd = static_cast<uint32_t>(available_space);
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +00001284}
1285
1286} // namespace cricket