How to calculate Round Trip Time in Ping utility implementation in UnetStack - unetstack

I had developed ping utility similar to the ping example available in UnetStack1.3 (/samples/ping) to ping a remote node over a multi-hop link, but unable to calculate the Round Trip Time (RTT), when I transmit ping packet using routing agent with static route information added to routing table using RouteDiscoveryNtf as there is no timing information avaiable in upper layer notifications (DatagramNtf or DatagramDeliveryNtf or DatagramFailureNtf).
Calculation of round trip time is the difference of rxtime and txtime available with TxFrameNtf and RxFrameNtf as implemented in clousure (fshrc.groovy) in the ping example.
I also tried analyzing the ping utility implemented in UnetStack3, but unable to makeout. Please let me know how the RTT is calculated.

Here's a simplified version of the implementation of the ping command in UnetStack3:
def ping(int n, int m = 3, long timeout = 30000) {
println "PING $n"
AgentID router = agentForService(Services.ROUTING)
int p = 0
m.times { count ->
def t0 = currentTimeMillis()
router << new DatagramReq(to: n, reliability: true)
def ntf = receive({
it instanceof DatagramDeliveryNtf || it instanceof DatagramFailureNtf
}, timeout)
def t = currentTimeMillis()-t0
if (ntf == null || ntf instanceof DatagramFailureNtf) {
println "Request timeout for seq $count"
} else {
p++
println "Response from $n: seq=$count rthops=2 time=$t ms"
}
delay(5000)
}
println "$m packets transmitted, $p packets received, ${Math.round(100*(m-p)/m)}% packet loss"
}

Related

using usocket seems to halt the loop (micropython)

I'm trying to code a simple program for a ESP32 board.
My main program is fairly simple and it has to run on a loop.
On the side, the device also needs to be able to respond to HTTP requests with a very simple response.
This is my attempt (a rework of https://randomnerdtutorials.com/micropython-esp32-esp8266-bme280-web-server/):
try:
import usocket as socket
except:
import socket
from micropython import const
import time
REFRESH_DELAY = const(60000) #millisecondi
def do_connect():
import network
wlan = network.WLAN(network.STA_IF)
wlan.active(True)
if not wlan.isconnected():
print('connecting to network...')
wlan.config(dhcp_hostname=HOST)
wlan.connect('SSID', 'PSWD')
while not wlan.isconnected():
pass
print('network config:', wlan.ifconfig())
import json
import esp
esp.osdebug(None)
import gc
gc.collect()
do_connect()
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.bind((HOST, SENSOR_SCKT_PORT))
s.listen(5)
prevRun = 0
i = 0
while True:
print("iteration #"+str(i))
i += 1
# run every 60 seconds
curRun = int(round(time.time() * 1000))
if curRun - prevRun >= REFRESH_DELAY:
prevRun = curRun
# MAIN PROGRAM
# ......
# whole bunch of code
# ....
# run continuously:
try:
if gc.mem_free() < 102000:
gc.collect()
conn, addr = s.accept()
conn.settimeout(3.0)
print('Got a connection from %s' % str(addr))
request = conn.recv(1024)
conn.settimeout(None)
request = str(request)
#print('Content = %s' % request)
measurements = 'some json stuff'
conn.send('HTTP/1.1 200 OK\n')
conn.send('Content-Type: text/html\n')
conn.send('Connection: close\n\n')
conn.send(measurements)
conn.close()
except OSError as e:
conn.close()
print('Connection closed')
what happens is I only get the iteration #0, and then the while True loop halts.
If I ping this server with a HTTP request, I get a correct response, AND the loop advances to iteration #1 and #2 (no idea why it thinks I pinged it with 2 requests).
So it seems that socket.listen(5) is halting the while loop.
Is there any way to avoid this?
Any other solution?
I don't think that threading is an option here.
The problem is that s.accept() is a blocking call...it won't return until it receives a connection. This is why it pauses your loop.
The easiest solution is probably to check whether or not a connection is waiting before calling s.accept(); you can do this using either select.select or select.poll. I prefer the select.poll API, which would end up looking something like this:
import esp
import gc
import json
import machine
import network
import select
import socket
import time
from micropython import const
HOST = '0.0.0.0'
SENSOR_SCKT_PORT = const(1234)
REFRESH_DELAY = const(60000) # milliseconds
def wait_for_connection():
print('waiting for connection...')
wlan = network.WLAN(network.STA_IF)
while not wlan.isconnected():
machine.idle()
print('...connected. network config:', wlan.ifconfig())
esp.osdebug(None)
gc.collect()
wait_for_connection()
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.bind((HOST, SENSOR_SCKT_PORT))
s.listen(5)
poll = select.poll()
poll.register(s, select.POLLIN)
prevRun = 0
i = 0
while True:
print("iteration #"+str(i))
i += 1
# run every 60 seconds
curRun = int(round(time.time() * 1000))
if curRun - prevRun >= REFRESH_DELAY:
prevRun = curRun
# MAIN PROGRAM
# ......
# whole bunch of code
# ....
# run continuously:
try:
if gc.mem_free() < 102000:
gc.collect()
events = poll.poll(100)
if events:
conn, addr = s.accept()
conn.settimeout(3.0)
print('Got a connection from %s' % str(addr))
request = conn.recv(1024)
conn.settimeout(None)
request = str(request)
# print('Content = %s' % request)
measurements = 'some json stuff'
conn.send('HTTP/1.1 200 OK\n')
conn.send('Content-Type: text/html\n')
conn.send('Connection: close\n\n')
conn.send(measurements)
conn.close()
except OSError:
conn.close()
print('Connection closed')
You'll note that I've taken a few liberties with your code to get it running on my device and to appease my sense of style; primarily, I've excised most of your do_connect method and put all the imports at the top of the file.
The only real changes are:
We create a select.poll() object:
poll = select.poll()
We ask it to monitor the s variable for POLLIN events:
poll.register(s, select.POLLIN)
We check if any connections are pending before attempting to handle a connection:
events = poll.poll(100)
if events:
conn, addr = s.accept()
conn.settimeout(3.0)
[...]
With these changes in place, running your code and making a request looks something like this:
iteration #0
iteration #1
iteration #2
iteration #3
iteration #4
iteration #5
iteration #6
Got a connection from ('192.168.1.169', 54392)
iteration #7
iteration #8
iteration #9
iteration #10
Note that as written here, your loop will iterate at least once every 100ms (and you can control that by changing the timeout on our call to poll.poll()).
Note: the above was tested on an esp8266 device (A Wemos D1 clone) running MicroPython v1.13-268-gf7aafc062).

how to set coreSize parameter in Hystrix?

I'm working on a spring boot project about electronic contract recently.And There has an interface of raiseContract().Considering that the traffic of this interface will be large in the future,My Leader let me use Hystrix to defender it.And I did not use it before.I am learning it and trying to use it on the interface.I use ThreadPool Isolation Strategy and I don't konw how to set
the parameter of coreSize reasonable in ThreadPoolProperties.In other words,I want to know what should I follow to set it.
I did a lot of research,but I did not get the answer.All of Answer is about the meaning of coreSize,maxQueueSize etc.
Here is my code:
#HystrixCommand(
groupKey = "contractGroup",
commandKey = "raiseContract",
fallbackMethod = "raiseContractFallback",
threadPoolProperties = {
#HystrixProperty(name = "coreSize", value = "20"),
#HystrixProperty(name = "maxQueueSize", value = "150"),
#HystrixProperty(name = "queueSizeRejectionThreshold", value = "100")},
commandProperties = {
#HystrixProperty(name = "execution.isolation.thread.timeoutInMilliseconds", value = "15000"),
#HystrixProperty(name = "circuitBreaker.requestVolumeThreshold", value = "5"),
#HystrixProperty(name = "circuitBreaker.errorThresholdPercentage", value = "50"),
#HystrixProperty(name = "circuitBreaker.sleepWindowInMilliseconds", value = "3000"),
#HystrixProperty(name = "fallback.isolation.semaphore.maxConcurrentRequests", value = "20")
})
As you are already aware of, there are 3 main threadPool configurations:
coreSize: Number of threads that will be maintained in the pool
maxSize: Defines how much extra threads are allowed in case need arises.
maxQueueSize: Queue size of the tasks
Now lets start with an example. Assume there is a service using hystrix, call it HystrixService, for which coreSize = maxSize = n and maxQueueSize = -1 (default case). This means at a time at most 'n' tasks will be executed. Any extra task that is coming will be rejected (fallback will be executed).
So, in ideal scenario, you have to ensure that this HystrixService doesn't reject any request coming to it. You need to know at max how many requests can there be on HystrixService. So if throughput on HystrixService is 10 Requests per second, then max concurrent requests on HystrixService can be 10. Now suppose latency of HystrixService is 2 sec, then by the time it responds to first 10 requests, 10 more requests will come. i.e. total requests = 2 * 10 = 20. So coreSize in this case should be 20.
This is same as mentioned in hystrix documentation,
coreSize = Peak Request per sec × P99 latency + some breathing room
Now, you can keep maxSize and queueSize a bit high, so that it doesn't reject requests, in case there are sudden throughput spikes on your service.

Unable to transcode from audio/l16;rate=48000;channel=1 to one of: audio/x-float-array; rate=16000; channels=1,

I am currently working on Softbanks' robot Pepper and I try to use Watson speech-to-text solution on Pepper's audio buffers remote streaming by using websocket protocol.
I used the answer to that former question NAO robot remote audio problems to find a way to access remotly pepper's audio buffers and that project https://github.com/ibm-dev/watson-streaming-stt to learn how to use websocket protocole to use watson streaming stt.
However, after I open my websocket application, I start sending buffers to watson and after a few sendings, I receive error: 'Unable to transcode from audio/l16;rate=48000;channel=1 to one of: audio/x-float-array; rate=16000; channels=1'
Each time I'm trying to send Pepper's audio buffer to watson, it is unable to understand it.
I compared data I send with data sent in watson streaming stt example (using pyaudio streaming from microphone instead of Pepper's buffer streaming) and I don't see any difference. Both time I'm pretty sure that I am sending a string containing raw chunks of bytes. Which is what Watson asks for in it documentation.
I try to send chunks of 8192 bytes with a sample rate of 48kHz and I can easily convert Pepper's audio buffer in hexa so I don't understand why Watson can't transcode it.
Here is my code:
# -*- coding: utf-8 -*-
#!/usr/bin/env python
import argparse
import base64
import configparser
import json
import threading
import time
from optparse import OptionParser
import naoqi
import numpy as np
import sys
from threading import Thread
import ssl
import websocket
from websocket._abnf import ABNF
CHANNELS = 1
NAO_IP = "172.20.10.12"
class SoundReceiverModule(naoqi.ALModule):
"""
Use this object to get call back from the ALMemory of the naoqi world.
Your callback needs to be a method with two parameter (variable name, value).
"""
def __init__( self, strModuleName, strNaoIp):
try:
naoqi.ALModule.__init__(self, strModuleName );
self.BIND_PYTHON( self.getName(),"callback" );
self.strNaoIp = strNaoIp;
self.outfile = None;
self.aOutfile = [None]*(4-1); # ASSUME max nbr channels = 4
self.FINALS = []
self.RECORD_SECONDS = 20
self.ws_open = False
self.ws_listening = ""
# init data for websocket interfaces
self.headers = {}
self.userpass = "" #userpass and password
self.headers["Authorization"] = "Basic " + base64.b64encode(
self.userpass.encode()).decode()
self.url = ("wss://stream.watsonplatform.net//speech-to-text/api/v1/recognize"
"?model=fr-FR_BroadbandModel")
except BaseException, err:
print( "ERR: abcdk.naoqitools.SoundReceiverModule: loading error: %s" % str(err) );
# __init__ - end
def __del__( self ):
print( "INF: abcdk.SoundReceiverModule.__del__: cleaning everything" );
self.stop();
def start( self ):
audio = naoqi.ALProxy( "ALAudioDevice", self.strNaoIp, 9559 );
self.nNbrChannelFlag = 3; # ALL_Channels: 0, AL::LEFTCHANNEL: 1, AL::RIGHTCHANNEL: 2; AL::FRONTCHANNEL: 3 or AL::REARCHANNEL: 4.
self.nDeinterleave = 0;
self.nSampleRate = 48000;
audio.setClientPreferences( self.getName(), self.nSampleRate, self.nNbrChannelFlag, self.nDeinterleave ); # setting same as default generate a bug !?!
audio.subscribe( self.getName() );
#openning websocket app
self._ws = websocket.WebSocketApp(self.url,
header=self.headers,
on_open = self.on_open,
on_message=self.on_message,
on_error=self.on_error,
on_close=self.on_close)
sslopt={"cert_reqs": ssl.CERT_NONE}
threading.Thread(target=self._ws.run_forever, kwargs = {'sslopt':sslopt}).start()
print( "INF: SoundReceiver: started!" );
def stop( self ):
print( "INF: SoundReceiver: stopping..." );
audio = naoqi.ALProxy( "ALAudioDevice", self.strNaoIp, 9559 );
audio.unsubscribe( self.getName() );
print( "INF: SoundReceiver: stopped!" );
print "INF: WebSocket: closing..."
data = {"action": "stop"}
self._ws.send(json.dumps(data).encode('utf8'))
# ... which we need to wait for before we shutdown the websocket
time.sleep(1)
self._ws.close()
print "INF: WebSocket: closed"
if( self.outfile != None ):
self.outfile.close();
def processRemote( self, nbOfChannels, nbrOfSamplesByChannel, aTimeStamp, buffer ):
"""
This is THE method that receives all the sound buffers from the "ALAudioDevice" module"""
print "receiving buffer"
# self.data_to_send = self.data_to_send + buffer
# print len(self.data_to_send)
#self.data_to_send = ''.join( [ "%02X " % ord( x ) for x in buffer ] ).strip()
self.data_to_send = buffer
#print("buffer type :", type(data))
#print("buffer :", buffer)
#~ print( "process!" );
print( "processRemote: %s, %s, %s, lendata: %s, data0: %s (0x%x), data1: %s (0x%x)" % (nbOfChannels, nbrOfSamplesByChannel, aTimeStamp, len(buffer), buffer[0],ord(buffer[0]),buffer[1],ord(buffer[1])) );
if self.ws_open == True and self.ws_listening == True:
print "sending data"
self._ws.send(self.data_to_send, ABNF.OPCODE_BINARY)
print "data sent"
#print self.data_to_send
aSoundDataInterlaced = np.fromstring( str(buffer), dtype=np.int16 );
#
aSoundData = np.reshape( aSoundDataInterlaced, (nbOfChannels, nbrOfSamplesByChannel), 'F' );
# print "processRemote over"
# processRemote - end
def on_message(self, ws, msg):
print("message")
data = json.loads(msg)
print data
if "state" in data:
if data["state"] == "listening":
self.ws_listening = True
if "results" in data:
if data["results"][0]["final"]:
self.FINALS.append(data)
# This prints out the current fragment that we are working on
print(data['results'][0]['alternatives'][0]['transcript'])
def on_error(self, ws, error):
"""Print any errors."""
print(error)
def on_close(self, ws):
"""Upon close, print the complete and final transcript."""
transcript = "".join([x['results'][0]['alternatives'][0]['transcript']
for x in self.FINALS])
print("transcript :", transcript)
self.ws_open = False
def on_open(self, ws):
"""Triggered as soon a we have an active connection."""
# args = self._ws.args
print "INF: WebSocket: opening"
data = {
"action": "start",
# this means we get to send it straight raw sampling
"content-type": "audio/l16;rate=%d;channel=1" % self.nSampleRate,
"continuous": True,
"interim_results": True,
# "inactivity_timeout": 5, # in order to use this effectively
# you need other tests to handle what happens if the socket is
# closed by the server.
"word_confidence": True,
"timestamps": True,
"max_alternatives": 3
}
# Send the initial control message which sets expectations for the
# binary stream that follows:
self._ws.send(json.dumps(data).encode('utf8'))
# Spin off a dedicated thread where we are going to read and
# stream out audio.
print "INF: WebSocket: opened"
self.ws_open = True
def version( self ):
return "0.6";
def main():
"""initialisation
"""
parser = OptionParser()
parser.add_option("--pip",
help="Parent broker port. The IP address or your robot",
dest="pip")
parser.add_option("--pport",
help="Parent broker port. The port NAOqi is listening to",
dest="pport",
type="int")
parser.set_defaults(
pip=NAO_IP,
pport=9559)
(opts, args_) = parser.parse_args()
pip = opts.pip
pport = opts.pport
# We need this broker to be able to construct
# NAOqi modules and subscribe to other modules
# The broker must stay alive until the program exists
myBroker = naoqi.ALBroker("myBroker",
"0.0.0.0", # listen to anyone
0, # find a free port and use it
pip, # parent broker IP
pport) # parent broker port
"""fin initialisation
"""
global SoundReceiver
SoundReceiver = SoundReceiverModule("SoundReceiver", pip) #thread1
SoundReceiver.start()
try:
while True:
time.sleep(1)
print "hello"
except KeyboardInterrupt:
print "Interrupted by user, shutting down"
myBroker.shutdown()
SoundReceiver.stop()
sys.exit(0)
if __name__ == "__main__":
main()
I would be thankful if anyone had any idea on how to bypass that error or on what to try to get useful info. I first believed that I was sending "wrong" data to watson however after lots of attempts I have no clue on how to fix that problem.
Thank you a lot,
Alex

Python3 TypeError: sequence item 0: expected a bytes-like object, int found

I'm trying to send an array over TCP from a server-like script to a client-like one. The array is variable, so the data is sent using packets and then joined together at the client.
The data I'm trying to send is from the MNIST hand-written digits dataset for Deep Learning. The server-side code is:
tcp = '127.0.0.1'
port = 1234
buffer_size = 4096
(X_train, y_train), (X_test, y_test) = mnist.load_data()
test_data = (X_test, y_test)
# Client-side Deep Learning stuff
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.connect((tcp, port))
x = pickle.dumps(test_data)
s.sendall(x)
s.close()
The client-side script loads a Neural Network that uses the test data to predict classes. The script for listening to said data is:
tcp = '127.0.0.1'
port = 1234
buffer_size = 4096
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.bind((tcp, port))
print ('Listening...')
s.listen(1)
conn, addr = s.accept()
data_arr = []
while True:
data_pack = conn.recv(buffer_size)
if not data: break
data_pack += data
my_pickle = b"".join(data_pack)
test_data = pickle.loads(my_pickle)
print ("Received: " + test_data)
conn.close()
# Irrelevant Deep Learning stuff...
The server sends the data without a hitch, but the client crashes when trying to join the packets received by the client (my_pickle = ...) with the following error:
TypeError: sequence item 0: expected a bytes-like object, int found
How should I format the join in order to recreate the data sent and use it for the rest of the script?
I ended up using both Pickle and ZeroMQ to handle the comunication protocol. An advantage of this method is that I can send more than one data package.
On the client side:
ip = '127.0.0.1'
port = '1234'
# ZeroMQ context
context = zmq.Context()
# Setting up protocol (client)
sock = context.socket(zmq.REQ)
sock.bind('tcp://'+ip+':'+port)
print('Waiting for connection at tcp://'+ip+':'+port+'...')
sock.send(pickle.dumps(X_send))
X_answer = sock.recv()
sock.send(pickle.dumps(y_send))
print('Data sent. Waiting for classification...')
y_answer = sock.recv()
print('Done.')
And on the server side:
# ZeroMQ Context
context = zmq.Context()
# Setting up protocol (server)
sock = context.socket(zmq.REP)
ip = '127.0.0.1'
port = '1234'
sock.connect('tcp://'+ip+':'+port)
print('Listening to tcp://'+ip+':'+port+'...')
X_message = sock.recv()
X_test = pickle.loads(X_message)
sock.send(pickle.dumps(X_message))
y_message = sock.recv()
y_test = pickle.loads(y_message)
print('Data received. Starting classification...')
# Classification process
sock.send(pickle.dumps(y_message))
print('Done.')

OpenFlow - How are ICMP messages handled

I am running a Ryu controller and a Mininet instance with 2 hosts and 1 switch like below.
H1---S---H2
Code in Ryu controller
from ryu.base import app_manager
from ryu.controller import ofp_event
from ryu.controller.handler import CONFIG_DISPATCHER, MAIN_DISPATCHER
from ryu.controller.handler import set_ev_cls
from ryu.ofproto import ofproto_v1_3
from ryu.lib.packet import packet
from ryu.lib.packet import ethernet
from ryu.lib.packet import ether_types
class SimpleSwitch13(app_manager.RyuApp):
OFP_VERSIONS = [ofproto_v1_3.OFP_VERSION]
def __init__(self, *args, **kwargs):
super(SimpleSwitch13, self).__init__(*args, **kwargs)
self.mac_to_port = {}
#set_ev_cls(ofp_event.EventOFPSwitchFeatures, CONFIG_DISPATCHER)
def switch_features_handler(self, ev):
datapath = ev.msg.datapath
ofproto = datapath.ofproto
parser = datapath.ofproto_parser
Basically the switch flow table is empty. In this case, when I run h1 ping h2 from my mininet console and record the packet exchanges, this is what I get in wireshark from host h1.
There is no router in the mininet instance. How am I receiving an ICMP Host Destination Unreachable Message from the same host that initiated the ping?
The app code you posted is not complete.
For complete simple_switch_13.py, you can get it from the osrg github.
Take a look, it is like this:
class SimpleSwitch13(app_manager.RyuApp):
OFP_VERSIONS = [ofproto_v1_3.OFP_VERSION]
def __init__(self, *args, **kwargs):
super(SimpleSwitch13, self).__init__(*args, **kwargs)
self.mac_to_port = {}
#set_ev_cls(ofp_event.EventOFPSwitchFeatures, CONFIG_DISPATCHER)
def switch_features_handler(self, ev):
datapath = ev.msg.datapath
ofproto = datapath.ofproto
parser = datapath.ofproto_parser
match = parser.OFPMatch()
actions = [parser.OFPActionOutput(ofproto.OFPP_CONTROLLER,
ofproto.OFPCML_NO_BUFFER)]
self.add_flow(datapath, 0, match, actions)
def add_flow(self, datapath, priority, match, actions, buffer_id=None):
ofproto = datapath.ofproto
parser = datapath.ofproto_parser
inst = [parser.OFPInstructionActions(ofproto.OFPIT_APPLY_ACTIONS,
actions)]
if buffer_id:
mod = parser.OFPFlowMod(datapath=datapath, buffer_id=buffer_id,
priority=priority, match=match,
instructions=inst)
else:
mod = parser.OFPFlowMod(datapath=datapath, priority=priority,
match=match, instructions=inst)
datapath.send_msg(mod)
#set_ev_cls(ofp_event.EventOFPPacketIn, MAIN_DISPATCHER)
def _packet_in_handler(self, ev):
# If you hit this you might want to increase
# the "miss_send_length" of your switch
if ev.msg.msg_len < ev.msg.total_len:
self.logger.debug("packet truncated: only %s of %s bytes",
ev.msg.msg_len, ev.msg.total_len)
msg = ev.msg
datapath = msg.datapath
ofproto = datapath.ofproto
parser = datapath.ofproto_parser
in_port = msg.match['in_port']
pkt = packet.Packet(msg.data)
eth = pkt.get_protocols(ethernet.ethernet)[0]
if eth.ethertype == ether_types.ETH_TYPE_LLDP:
# ignore lldp packet
return
dst = eth.dst
src = eth.src
dpid = datapath.id
self.mac_to_port.setdefault(dpid, {})
self.logger.info("packet in %s %s %s %s", dpid, src, dst, in_port)
# learn a mac address to avoid FLOOD next time.
self.mac_to_port[dpid][src] = in_port
if dst in self.mac_to_port[dpid]:
out_port = self.mac_to_port[dpid][dst]
else:
out_port = ofproto.OFPP_FLOOD
actions = [parser.OFPActionOutput(out_port)]
# install a flow to avoid packet_in next time
if out_port != ofproto.OFPP_FLOOD:
match = parser.OFPMatch(in_port=in_port, eth_dst=dst)
# verify if we have a valid buffer_id, if yes avoid to send both
# flow_mod & packet_out
if msg.buffer_id != ofproto.OFP_NO_BUFFER:
self.add_flow(datapath, 1, match, actions, msg.buffer_id)
return
else:
self.add_flow(datapath, 1, match, actions)
data = None
if msg.buffer_id == ofproto.OFP_NO_BUFFER:
data = msg.data
out = parser.OFPPacketOut(datapath=datapath, buffer_id=msg.buffer_id,
in_port=in_port, actions=actions, data=data)
datapath.send_msg(out)
This simple_switch_13.py app only handles layer 2 forwarding, which is your case.
As you can see, after the connection established, the switch_features_handler will listen on this event and add a send all flow to controller flow on the switch.(table-miss flow)
And for the normal states, when the controller receives PACKET_IN, it will check if the dst_MAC is in the mac_to_port. If yes, then output to the port, and at the same time insert a flow(whose match field is inport and dst_MAC); else(not in the array), the action is set to be FLOOD by assigning the outport=FLOOD.
That's the case in Layer 2 switching.
For ICMP messages handling in layer 3 switching, you need to read the rest_router.py code, which is a lot more complicated.
You get ICMP Host Destination Unreachable because the ARP request is never answered by h2.
Since h1 gets no ARP reply, ICMP error message comes from its own IP stack.

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