Add even more of the source

This should be about everything needed to build so far?

StreamingLoadTool/Nokia_N90.conf

@@ -0,0 +1,127 @@
+# This is a StreamingLoadTool config file
+
+#player user agent name
+player QTS
+
+# Use the "clienttype" directive to specify whether StreamingLoadTool should make
+# RTSP / UDP connections or RTSP / HTTP connections or . Say "http" for
+# the latter, "udp" for the former. Say "reliableudp" for reliable UDP.
+# Say "tcp" for straight interleaved RTSP / RTP
+clienttype udp
+
+# If doing RTSP / HTTP, set droppost to "yes" if you would like StreamingLoadTool
+# to drop the POST half of each RTSP / HTTP connection after sending the
+# PLAY. "yes" best emulates the "real" client behavior.
+droppost yes
+
+# Set this to the # of concurrent clients you would like StreamingLoadTool to maintain(default = 1)
+# When not running forever, this is also the number of clients to load before exiting
+concurrentclients	1
+
+# Specify a connection port for each connection(default 554)
+port		554
+
+# Specify a proxy IP address in dotted-decimal form. If 0, StreamingLoadTool
+# will not use a proxy to connect
+proxyip		0
+
+# Client window (size of UDP socket buffers). Default = 32768
+# For reliable UDP, this affects packet loss and the server's retransmission algorithm
+clientwindow 32768
+
+# StreamingLoadTool should send a TEARDOWN after streaming for this # of seconds
+movielength	60
+
+# If runforever is set to "no", StreamingLoadTool will quit after finishing the
+# list of URLs provided below. If "yes" StreamingLoadTool will loop forever. (Default = no)
+runforever no
+
+# Each instance of StreamingLoadTool must have a unique httpcookie value. This
+# can also be specified on the command-line (see usage by doing StreamingLoadTool -v) (Default = 1)
+httpcookie	1
+
+# Set to "yes" if you would like StreamingLoadTool to generate a connection log
+shouldlog	yes
+
+# Append junk data after each DESCRIBE request
+appendjunk	no
+
+# Location to place the connection log
+logpath		streamingloadtool.log
+
+# Interval in milliseconds between attempts to read media data. For acking
+# udp clients, this is also the interval between acks. (Default = 50 milliseconds)
+readinterval 10
+
+# how late should packets be allowed to be sent? Value is in seconds.
+# 0 = no late tolerance will be specified at all
+latetolerance 0
+
+# The "Packet-Range" header is used as an alternative to the standard "Range"
+# header on a Play request for specifying a range of packets. Leave
+# this line blank to issue a standard play, specify a packet range header
+# here to send that instead of standard Range header.
+packetplayheader
+
+# The overbuffer window size is the number of K bytes the server can send
+# ahead of time. This applies to all transports except "udp".
+overbufferwindowsize 5192
+
+# Set this to be the value of the x-RTP-Meta-Info header sent to the server.
+# If it is empty, no x-RTP-Meta-Info header will be sent. Otherwise, specify
+# the fields you would like to receive
+# rtpmetainfo tt;ft;pn;fd;md;sq
+
+# Set this to be the speed you want the streams at (1 = normal speed, 2 = 2x normal speed, etc; float)
+speed	1
+
+# Enable this to have StreamingLoadTool randomly thumb around
+randomthumb no
+
+# Set sendoptions "yes" to send an OPTIONS request before executing each DESCRIBE request
+sendoptions no
+
+# Set requestrandomdata "yes" to send an OPTIONS request with a random data request header before executing each DESCRIBE request
+requestrandomdata no
+
+# How many random bytes the server should send in the body of the OPTIONS response
+randomdatasize 0
+
+# How often should the client send RTCP messages in milliseconds. (Default = 5000; ACK's in reliableudp are sent ASAP)
+rtcpinterval 1000
+
+# List of rtsp URLs for StreamingLoadTool to execute
+url rtsp://foo.bar.com/sample.mov
+
+# List of users:passwords for authentication (format: user:password)
+user user0:pass0
+
+#The advertised bandwidth in bps (default: not sent)
+bandwidth 31235
+
+#The player buffer space per stream in bytes. Default of 0 is unlimited
+buffer 15000
+
+#target delay in milliseconds.  Default 3000.  Use 0 to turn it off
+#This is the desired amount of playback time to keep in the buffer.
+delay 10000
+
+#Start play time delay; expressed as a fraction of the target delay.  This is how much data there should be in the buffer before
+#the media starts playing.  Use 0.0 to start playing immediately, and 1.0 to start playing when the target delay has been met. Default = 5.0
+startdelay 0.5
+
+#Turn on 3GPP features and headers? 3GPP should be not be enabled on reliableudp transport; default = no
+enable3GPP yes
+
+# The advertised guarenteed bit rate(GBW), maximum bit rate(MBW), and maximum transfer delay(MTD) for the wireless link
+# Units are in kilobits per seconds and milliseconds.
+# Default = 0 = no specified value
+#GBW 32
+#MBW 128
+#MTD 2000
+
+# Enable a forced PlayoutDelay value for 3GPP RTCP packets (The value defined for the playoutDelay preference will be sent)
+enableForcePlayoutDelay no
+
+# PlayoutDelay value to use if enableForcePlayoutDelay is enabled
+playoutDelay 65535

StreamingLoadTool/dummynet_fixed_start.script

@@ -0,0 +1,31 @@
+#!/bin/sh
+#
+#
+echo "SCRIPT $0 [ip address] "
+echo "TEST Set a limited bandwidth."
+echo ""
+
+ADDRESS="127.0.0.1"
+
+if [ ${1} ]; then
+    ADDRESS=${1}
+    echo "Address set. Using $ADDRESS"
+ else
+    echo "No Address given. Using $ADDRESS"
+fi
+
+ipfw delete pipe 1
+ipfw delete pipe 2
+ipfw delete set 11
+ipfw delete set 12
+
+ipfw pipe 1 config bw 80kbits/s delay 0 queue 100 noerror
+ipfw pipe 2 config bw 80kbits/s delay 0 queue 100 noerror
+
+ipfw add 11 set 11 pipe 1 src-ip $ADDRESS proto udp in
+ipfw add 12 set 12 pipe 2 src-ip $ADDRESS proto udp out
+
+ipfw set enable 11
+ipfw set enable 12
+ipfw enable firewall one_pass
+

StreamingLoadTool/dummynet_stop.script

@@ -0,0 +1,10 @@
+echo "SCRIPT $0 "
+echo "TEST delete pipes, sets, and disable dummynet rule"
+
+ipfw delete pipe 1
+ipfw delete pipe 2
+ipfw delete set 11
+ipfw delete set 12
+
+ipfw disable firewall one_pass
+

StreamingLoadTool/dummynet_variable.script

@@ -0,0 +1,73 @@
+#!/bin/sh
+#
+#
+echo "SCRIPT $0 [ip address] [down Kbits] [up Kbits] [test seconds] "
+echo "TEST Alternate between no bandwidth limits and limited bandwidth with a delay between"
+echo ""
+
+ADDRESS="127.0.0.1"
+UP="32"
+DOWN="100"
+SLEEP="30"
+
+if [ ${1} ]; then
+    ADDRESS=${1}
+    echo "Address set. Using $ADDRESS"
+ else
+    echo "No Address given. Using $ADDRESS"
+fi
+
+if [ ${2} ]; then
+    DOWN=${2}
+    echo "Down link limit set. Using "$DOWN"kbits/s"
+else
+    echo "No down link limit given. Using "$DOWN"kbits/s"
+    
+fi
+
+if [ ${3} ]; then
+    UP=${3}
+    echo "Up link limit set. Using "$UP"kbits/s"
+else
+    echo "No up link limit given. Using "$UP"kbits/s"
+fi
+
+
+if [ ${4} ]; then
+    SLEEP=${4}
+    echo "Test time set. Using $SLEEP seconds"
+else
+    echo "No test time. Using $SLEEP seconds"
+fi
+
+echo "Starting test"
+
+while [ 1 ]
+ do 
+    
+    ipfw delete pipe 1 2
+    ipfw delete set 11 12
+    ipfw disable firewall
+    echo ""
+    echo "Bandwidth open - no limit set"
+    echo "Sleeping $SLEEP seconds"
+    sleep 30
+    echo ""
+
+    echo "Setting $DOWN kbits down and $UP kbits up"
+    ipfw pipe 1 config bw "$DOWN"kbits/s delay 0 queue 100 noerror
+    ipfw pipe 2 config bw "$UP"kbits/s delay 0 queue 100 noerror
+
+    ipfw add 11 set 11 pipe 1 src-ip $ADDRESS proto udp in
+    ipfw add 12 set 12 pipe 2 src-ip $ADDRESS proto udp out
+
+    ipfw set enable 11 
+    ipfw set enable 12
+    ipfw enable firewall one_pass
+
+    echo "Sleeping $SLEEP seconds"
+    sleep 30
+    
+
+done
+

StreamingLoadTool/gdb_script

@@ -0,0 +1,6 @@
+dir ../APICommonCode
+dir ../CommonUtilitiesLib
+dir ../OSMemoryLib
+dir ../PrefsSourceLib
+dir ../RTSPClientLib
+handle SIGPIPE nopass nostop noprint

StreamingLoadTool/gdbspam

@@ -0,0 +1 @@
+gdb ./StreamingLoadTool -x gdb_script

StreamingLoadTool/parseSLTOut.py

@@ -0,0 +1,203 @@
+#!python
+
+"""Script for parsing the output of a StreamingLoadTool file.  Can be used both as a script and as a python module.
+The output is a commma separated file.  The StreamingLoadTool has to be run with -V 3 or above.
+
+Usage: ParseSLTOut.py [-h] [-b interval] [-s all] [filename]
+
+options:
+	-h                show this help message and exit
+	-b                Take the packet lengths and categorizes them according to their arrival time.
+	                  The packet lengths within the same bucket (of size interval seconds) are summed up
+	                  and averaged over the interval, which is useful for seeing the bit rate.
+	-s                The streams to parse: one of video, audio, all; default is video
+
+If filename is missing, then the script will read from standard in.  The script writes to standard out."""
+
+"""
+ *
+ * @APPLE_LICENSE_HEADER_START@
+ *
+ * Copyright (c) 1999-2008 Apple Inc.  All Rights Reserved.
+ *
+ * This file contains Original Code and/or Modifications of Original Code
+ * as defined in and that are subject to the Apple Public Source License
+ * Version 2.0 (the 'License'). You may not use this file except in
+ * compliance with the License. Please obtain a copy of the License at
+ * http://www.opensource.apple.com/apsl/ and read it before using this
+ * file.
+ * 
+ * The Original Code and all software distributed under the License are
+ * distributed on an 'AS IS' basis, WITHOUT WARRANTY OF ANY KIND, EITHER
+ * EXPRESS OR IMPLIED, AND APPLE HEREBY DISCLAIMS ALL SUCH WARRANTIES,
+ * INCLUDING WITHOUT LIMITATION, ANY WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE, QUIET ENJOYMENT OR NON-INFRINGEMENT.
+ * Please see the License for the specific language governing rights and
+ * limitations under the License.
+ * 
+ * @APPLE_LICENSE_HEADER_END@
+ *
+"""
+
+import re, sys
+
+#the regular expression for matching lines in the SLT output
+playRE = re.compile(\
+		r"^Receiving track (\d+), trackID=(\d+), (\w+) at time (\d+)")
+timeRE = re.compile(\
+		r"^Processing incoming packets at time (\d+)")
+processedRE = re.compile(\
+		r"^Processed packet: track=(\d+), len=(\d+), seq=(\d+), time=(\d+)\((\d+)\); bufferingDelay=(\d+), FBS=(\d+)")
+
+
+def parseSLTInput(inputFile):
+	"""
+	Parses a string consisting of outputs from StreamingLoadTool.
+	The output is a list, where the index is the track index.  The value is a map that maps from processing time to
+	packets.  Each packet is a 3-tuple containing the packet length(in bytes), the sequence number, and the timestamp.
+	"""
+	videoTrackIndex = None
+	audioTrackIndex = None
+	localTimeBase = None
+	curTime = None
+
+	packetTable = [{}, {}]
+
+	for line in inputFile:
+		matchObj = playRE.match(line)
+		if matchObj is not None:
+			trackIndex, trackID, trackType, startTime = matchObj.groups();
+			if trackType == "video":
+				videoTrackIndex = int(trackIndex)
+			elif trackType == "audio":
+				audioTrackIndex = int(trackIndex)
+			localTimeBase = int(startTime)
+			continue
+
+		matchObj = timeRE.match(line)
+		if matchObj is not None:
+			processingTime, = matchObj.groups()
+			curTime = int(processingTime)
+			continue
+
+		matchObj = processedRE.match(line)
+		if matchObj is not None:
+			trackIndex, packetLen, seqNum, timeStamp, timeStampInMediaTime, playoutDelay, freeBufferSpace = matchObj.groups()
+			trackIndex = int(trackIndex);
+			while trackIndex >= len(packetTable):
+				packetTable.append({})
+			packetList = packetTable[trackIndex].setdefault(curTime, [])
+			packetList.append( (int(packetLen), int(seqNum), int(timeStamp)) )
+			continue
+
+	if localTimeBase is None:
+		sys.exit("Parse error: cannot find a track")
+
+	# modify the processing times to be 0-based
+	newPacketTable = [{}] * len(packetTable)
+	for trackIndex in range(len(packetTable)):
+		newPackets = {}
+		for (curTime, packet) in packetTable[trackIndex].items():
+			curTime -= localTimeBase
+			newPackets[curTime] = packet
+		packetTable[trackIndex] = newPackets
+
+	return (packetTable, videoTrackIndex, audioTrackIndex)
+
+
+def calcBitRate(xList, yList, interval):
+	"""xList is a list of processing time, and yList is a list of packet size.
+	The function will calculate the bitrate, divided into interval-sized buckets, and returns it as a list
+	of (processing time, bitrate) pair.
+	xList is expected to be a list of time in milliseconds, and yList is expected to be a list of packet size in bytes."""
+	maxIndex = max(xList) / interval
+	bitsReceived = [0] * (maxIndex + 1)
+
+	for i in range(len(yList)):
+		x = xList[i]
+		y = yList[i]
+
+		bitsReceived[x / interval] += y * 8
+	
+	return [ (i * interval, (bitsReceived[i] * 1000) / interval) for i in range(len(bitsReceived))]
+
+
+if __name__ == "__main__":
+	# the file is ran as a script
+
+	# first parse the command line
+	import getopt
+	from decimal import Decimal
+
+	try:
+		optlist, args = getopt.getopt(sys.argv[1:], "hb:s:")
+	except getopt.GetoptError:
+		sys.exit(__doc__)
+
+	categorize = False
+	streams = 'video'
+	for opt, optarg in optlist:
+		if opt == '-h':
+			print __doc__
+			sys.exit(0)
+		elif opt == '-b':
+			categorize = True
+			try:
+				interval = int(Decimal(optarg) * 1000)
+			except:
+				sys.exit(__doc__)
+		elif opt == '-s':
+			streams = optarg
+			if streams != 'video' and streams != 'audio' and streams != 'all':
+				sys.exit(__doc__)
+
+	if len(args) == 0:
+		inputFile = sys.stdin
+	elif len(args) == 1:
+		inputFile = open(args[0], 'r')
+	else:
+		sys.exit(__doc__)
+	
+	(packetTable, videoIndex, audioIndex) = parseSLTInput(inputFile)
+
+	if streams == 'video':
+		if videoIndex is None:
+			sys.exit("Parse error: Cannot find a video stream")
+		packets = packetTable[videoIndex]
+	elif streams == 'audio':
+		if audioIndex is None:
+			sys.exit("Parse error: Cannot find an audio stream")
+		packets = packetTable[audioIndex]
+	else:
+		packets = {}
+		for perStreamPackets in packetTable:
+			for (time, packetList) in perStreamPackets.items():
+				packets.setdefault(time, []).extend(packetList)
+
+	if len(packets) == 0:
+		sys.exit("Error: Cannot find a stream")
+
+
+	def millisecondsToSec(time):
+		"""Convert time to 123.456 form as a string"""
+		return str(time / 1000) + '.' + str(time % 1000)
+
+	if categorize:
+		xList = []
+		yList = []
+
+		for processingTime, packetList in packets.items():
+			for (packetLen, seqNum, timeStamp) in packetList:
+				xList.append(processingTime)
+				yList.append(packetLen)
+		data = calcBitRate(xList, yList, interval)
+		for (time, bitrate) in data:
+			print "%s, %i" % (millisecondsToSec(time), bitrate)
+	else:
+		processingTimes = packets.keys()
+		processingTimes.sort()
+		for processingTime in processingTimes:
+			packetList = packets[processingTime]
+			for (packetLen, seqNum, timeStamp) in packets[processingTime]:
+				# output.append( (processingTime, packetLen) )
+				print ("%s, %i") % ( millisecondsToSec(processingTime), packetLen )

StreamingLoadTool/streamingloadtool.cfg

@@ -0,0 +1,81 @@
+# This is a StreamingLoadTool config file
+
+# Use the "clienttype" directive to specify whether StreamingLoadTool should make
+# RTSP / UDP connections or RTSP / HTTP connections or . Say "http" for
+# the latter, "udp" for the former. Say "reliableudp" for reliable UDP.
+# Say "tcp" for straight interleaved RTSP / RTP
+clienttype reliableudp
+
+# If doing RTSP / HTTP, set droppost to "yes" if you would like StreamingLoadTool
+# to drop the POST half of each RTSP / HTTP connection after sending the
+# PLAY. "yes" best emulates the "real" client behavior.
+droppost yes
+
+# Set this to the # of concurrent clients you would like StreamingLoadTool to maintain
+concurrentclients	1
+
+# Specify a connection port for each connection
+port		554
+
+# Specify a proxy IP address in dotted-decimal form. If 0, StreamingLoadTool
+# will not use a proxy to connect
+proxyip		0
+
+# Client window (size of UDP socket buffers).
+# For reliable UDP, this affects packet loss and the server's
+# retransmission algorithm
+clientwindow 32768
+
+# StreamingLoadTool should send a TEARDOWN after streaming for this # of seconds
+movielength	40
+
+# If runforever is set to "no", StreamingLoadTool will quit after finishing the
+# list of URLs provided below. If "yes" StreamingLoadTool will loop forever.
+runforever	yes
+
+# Each instance of StreamingLoadTool must have a unique httpcookie value. This
+# can also be specified on the command-line (see usage by doing StreamingLoadTool -v)
+httpcookie	1
+
+# Set to "yes" if you would like StreamingLoadTool to generate a connection log
+shouldlog	yes
+
+# Append junk data after each DESCRIBE request
+appendjunk	no
+
+# Location to place the connection log
+logpath		streamingloadtool.log
+
+# Interval in milliseconds between attempts to read media data. For acking
+# udp clients, this is also the interval between acks.
+readinterval 10
+
+# how late should packets be allowed to be sent? Value is in seconds.
+# 0 = no late tolerance will be specified at all
+latetolerance 0
+
+# The "Packet-Range" header is used as an alternative to the standard "Range"
+# header on a Play request for specifying a range of packets. Leave
+# this line blank to issue a standard play, specify a packet range header
+# here to send that instead of standard Range header.
+packetplayheader
+
+# The overbuffer window size is the number of K bytes the server can send
+# ahead of time. This applies to all transports except "udp".
+overbufferwindowsize 5192
+
+# Set this to be the value of the x-RTP-Meta-Info header sent to the server.
+# If it is empty, no x-RTP-Meta-Info header will be sent. Otherwise, specify
+# the fields you would like to receive
+# rtpmetainfo tt;ft;pn;fd;md;sq
+
+# Set this to be the speed you want the streams at (1 = normal speed, 2 = 2x normal speed, etc)
+speed	1
+
+# Enable this to have StreamingLoadTool randomly thumb around
+randomthumb no
+
+# List of rtsp URLs for StreamingLoadTool to execute
+url rtsp://foo.bar.com/sample.mov
+
+

StreamingLoadTool/streamingloadtool.conf

@@ -0,0 +1,129 @@
+# This is a StreamingLoadTool config file
+
+#player user agent name
+player QTS
+
+# Use the "clienttype" directive to specify whether StreamingLoadTool should make
+# RTSP / UDP connections or RTSP / HTTP connections or . Say "http" for
+# the latter, "udp" for the former. Say "reliableudp" for reliable UDP.
+# Say "tcp" for straight interleaved RTSP / RTP
+# Use "3gudp" for 3gpp rate adaptation over UDP
+clienttype reliableudp
+
+# If doing RTSP / HTTP, set droppost to "yes" if you would like StreamingLoadTool
+# to drop the POST half of each RTSP / HTTP connection after sending the
+# PLAY. "yes" best emulates the "real" client behavior.
+droppost yes
+
+# Set this to the # of concurrent clients you would like StreamingLoadTool to maintain(default = 1)
+# When not running forever, this is also the number of clients to load before exiting
+concurrentclients	1
+
+# Specify a connection port for each connection(default 554)
+port		554
+
+# Specify a proxy IP address in dotted-decimal form. If 0, StreamingLoadTool
+# will not use a proxy to connect
+proxyip		0
+
+# Client window (size of UDP socket buffers). Default = 32768
+# For reliable UDP, this affects packet loss and the server's retransmission algorithm
+clientwindow 32768
+
+# StreamingLoadTool should send a TEARDOWN after streaming for this # of seconds
+movielength	40
+
+# If runforever is set to "no", StreamingLoadTool will quit after finishing the
+# list of URLs provided below. If "yes" StreamingLoadTool will loop forever. (Default = no)
+runforever yes
+
+# Each instance of StreamingLoadTool must have a unique httpcookie value. This
+# can also be specified on the command-line (see usage by doing StreamingLoadTool -v) (Default = 1)
+httpcookie	1
+
+# Set to "yes" if you would like StreamingLoadTool to generate a connection log
+shouldlog	yes
+
+# Append junk data after each DESCRIBE request
+appendjunk	no
+
+# Location to place the connection log
+logpath		streamingloadtool.log
+
+# Interval in milliseconds between attempts to read media data. For acking
+# udp clients, this is also the interval between acks. (Default = 50 milliseconds)
+readinterval 10
+
+# how late should packets be allowed to be sent? Value is in seconds.
+# 0 = no late tolerance will be specified at all
+latetolerance 0
+
+# The "Packet-Range" header is used as an alternative to the standard "Range"
+# header on a Play request for specifying a range of packets. Leave
+# this line blank to issue a standard play, specify a packet range header
+# here to send that instead of standard Range header.
+packetplayheader
+
+# The overbuffer window size is the number of K bytes the server can send
+# ahead of time. This applies to all transports except "udp".
+overbufferwindowsize 5192
+
+# Set this to be the value of the x-RTP-Meta-Info header sent to the server.
+# If it is empty, no x-RTP-Meta-Info header will be sent. Otherwise, specify
+# the fields you would like to receive
+# rtpmetainfo tt;ft;pn;fd;md;sq
+
+# Set this to be the speed you want the streams at (1 = normal speed, 2 = 2x normal speed, etc; float)
+speed	1
+
+# Enable this to have StreamingLoadTool randomly thumb around
+randomthumb no
+
+# Set sendoptions "yes" to send an OPTIONS request before executing each DESCRIBE request
+sendoptions no
+
+# Set requestrandomdata "yes" to send an OPTIONS request with a random data request header before executing each DESCRIBE request
+requestrandomdata no
+
+# How many random bytes the server should send in the body of the OPTIONS response
+randomdatasize 0
+
+# How often should the client send RTCP messages in milliseconds. (Default = 5000; ACK's in reliableudp are sent ASAP)
+rtcpinterval 1000
+
+# List of rtsp URLs for StreamingLoadTool to execute
+url rtsp://foo.bar.com/sample.mov
+
+# List of users:passwords for authentication (format: user:password)
+user user0:pass0
+
+#The advertised bandwidth in bps (default: not sent)
+bandwidth 50000
+
+#The player buffer space per stream in bytes. Default of 0 is unlimited
+buffer 100000
+
+#target delay in milliseconds.  Default 3000.  Use 0 to turn it off
+#This is the desired amount of playback time to keep in the buffer.
+delay 10000
+
+#Start play time delay; expressed as a fraction of the target delay.  This is how much data there should be in the buffer before
+#the media starts playing.  Use 0.0 to start playing immediately, and 1.0 to start playing when the target delay has been met. Default = 5.0
+#Use -1 for disabled.
+startdelay 0.5
+
+#Turn on 3GPP features and headers? 3GPP should be not be enabled on reliableudp transport; default = no
+enable3GPP no
+
+# The advertised guarenteed bit rate(GBW), maximum bit rate(MBW), and maximum transfer delay(MTD) for the wireless link
+# Units are in kilobits per seconds and milliseconds.
+# Default = 0 = no specified value
+#GBW 32
+#MBW 128
+#MTD 2000
+
+# Enable a forced PlayoutDelay value for 3GPP RTCP packets (The value defined for the playoutDelay preference will be sent)
+enableForcePlayoutDelay no
+
+# PlayoutDelay value to use if enableForcePlayoutDelay is enabled
+playoutDelay 65535