The rtptools distribution consists of a number of small applications that can be used for processing RTP data.
Sources for a variety of platforms and binaries for Windows are available from http://www.cs.columbia.edu/IRT/rtptools/software/download.
The RTP tools should compile on any Posix-compliant platform supporting sockets, as well as Windows/NT/95/98/2000 (Win32). They have been tested on SunOS 4.1, SunOS 5.x (Solaris), Linux, NT 4.0, SGI Irix, and HP-UX. Edit the directories and libraries at the top of Makefile and type make. The compiler must support ANSI C: gcc does, Sun's old /usr/ucb/cc does not.
Note: You must use the sun4 architecture for
SunOS 4.1.x and sun5 for SunOS 5.x (Solaris). You will get
system call errors if you do not.
For quite a character, who desire to compile on Borland C++ Builder, please open dump_bcb.bpr, play_bcb.bpr, send_bcb.bpr and trans_bcb.bpr under bcb directory. Only pressing ctr-F9 needed for compilation, and the tool will be generated on the same directory.
Network addresses can be either multicast or unicast addresses, unless stated otherwise. They may be specified in dotted-decimal notation (e.g., 224.2.0.1) or as a host name (e.g., lupus.fokus.gmd.de). Port numbers must be given as decimal numbers in the range of 1 to 65535. Network addresses are specified as destination/port/ttl. The time-to-live (ttl) value is optional and only applies to multicast.
For all commands, the flag -h or -? will print a short usage summary.
Unless otherwise noted, input is taken from stdin, and output sent to
stdout. The extension .rtp is suggested for files
generated in rtpdump -F dump format.
rtpplay reads RTP session data, recorded by rtpdump -F dump from either the file or stdin, if file is not specified, sending it to network address destination and port port with a time-to-live value of ttl.
If the flag -T is given, the
timing between packets corresponds to the arrival timing rather than the
RTP timestamps. Otherwise, for RTP data packets, the timing given by
the RTP timestamps is used, smoothing interarrival jitter and restoring
packet sequence. RTCP packets are still sent with their original
timing. This may cause the relative order of RTP and RTCP packets to be
changed.
The source port(localport) for outgoing packets can be set with the -s flag. A random port is chosen if this flag is not specified.
The whole file is played unless the begin or end times are specified. Times are measured in seconds and fractions from the beginning of the recording.
The RTP clock frequency is read from the profile file if
given; the default profile (RFC 1890) is used if not. The profile file
contains lines with two fields each: the first is the numeric payload
type, the second the clock frequency. The values read from the profile
file are silently ignored if the -T flag is used.
If you want to loop a particular file, it is easiest to put the
rtpplay command in a shell script.
The -v flag has rtpplay display the packets generated on stdout.
rtpplay uses the hsearch (3C) library,
which may not be available on all operating systems.
If file is specified, the file is used instead of the
network address. If no network address is given, file input is expected
from stdin. The file must have been recorded using the rtpdump
The recording duration is measured in minutes.
From each packet,
only the first bytes of the payload are dumped (only
applicable for "dump" and "hex" formats).
Supported formats are:
The version number indicates the file format version, not the version
of RTP tools used to generate the file. The current file format version
is 1.0.
This is followed by one binary header (RD_hdr_t) and one
RD_packet_t structure for each received packet. All fields are
in network byte order. The RTP and RTCP packets are recorded as-is.
rtpsend sends an RTP packet stream with configurable
parameters. This is intended to test RTP features. The RTP or RTCP
headers are read from a file, generated by hand, a test program or rtpdump (format "ascii").
rtpsend [-a] [-l]
[-s sourceport]
[-f file]
destination/port[/ttl]
Packets are sent with a time-to-live value ttl.
If data is read from a file instead of stdin, the
-l(loop) flag resends the same sequence of packets again and
again.
The source port(localport) for outgoing packets can be set with the
-s flag. A random port is chosen if this flag is not
specified.
If the -a flag is specified, rtpsend
includes a router alert IP option in RTCP packets. This is used by the
YESSIR resource reservation protoccol.
The file file contains the description of the packets to
be sent. Within the file, each entry starts with a time value, in
seconds, relative to the beginning of the trace. The time value must
appear at the beginning of a line, without white space. Within an RTP
or RTCP packet description, parameters may appear in any order, without
white space around the equal sign. Lines are continued with initial
white space on the next line. Comment lines start with #. Strings are
enclosed in quotation marks.
rtptrans RTP/RTCP packets arriving from one of the addresses to all
other addresses. Addresses can be a multicast or unicast. TTL values
for unicast addresses are ignored. (Actually, doesn't check whether
packets are RTP or not.)
Additionally, the translator can translate VAT packets into RTP
packets. VAT control packets are translated into RTCP SDES packets with
a CNAME and a NAME entry. However, this is only intended to be used in
the following configuration: VAT packets arriving on a multicast
connection are translated into RTP and sent over a unicast link. RTP
packets are not (yet) translated into VAT packets and and all packets
arriving on unicast links are not changed at all. Therefore, currently
mainly the following topology is supported: multicast VAT -> translator
-> unicast RTP; and on the way back it should lokk like this multicast
VAT <- translator <- unicast VAT. This means that the audio agent on
the unicast link should be able use both VAT and RTP.
The rtptools were written by Henning Schulzrinne, with
enhancements by Ping
Pan and Akira Tsukamoto.
rtptrans was written by Dorgham
Sisalem and enhanced by Steve
Casner.
dump format.
format
text/binary
description
dump
binary
dump in binary format, suitable for rtpplay. The format is as follows:
The file starts with
#!rtpplay1.0 address/port\n
typedef struct {
struct timeval start; /* start of recording (GMT) */
u_int32 source; /* network source (multicast address) */
u_int16 port; /* UDP port */
} RD_hdr_t;
typedef struct {
u_int16 length; /* length of packet, including this header (may
be smaller than plen if not whole packet recorded) */
u_int16 plen; /* actual header+payload length for RTP, 0 for RTCP */
u_int32 offset; /* milliseconds since the start of recording */
} RD_packet_t;
header
like "dump", but don't save audio/video payload
payload
only audio/video payload
ascii
text
parsed packets (default), suitable for rtpsend:
844525628.240592 RTP len=176 from=131.136.234.103:46196 v=2 p=0 x=0
cc=0 m=0 pt=5 (IDVI,1,8000) seq=28178 ts=954052737 ssrc=0x124e2b58
844525628.243123 RTCP len=128 from=139.88.27.43:53154
(RR ssrc=0x125bd36f p=0 count=1 len=7
(ssrc=bc64b658 fraction=0.503906 lost=4291428375 last_seq=308007791
jit=17987961 lsr=2003335488 dlsr=825440558)
)
(SDES p=0 count=1 len=23
(src=0x125bd36f CNAME="yywhy@139.88.27.43" NAME="Michael Baldizzi
(NASA LeRC)" TOOL="vat-4.0a8" EMAIL="mbaldizzi@lerc.nasa.gov" )
)
hex
like ascii, but with hex dump of payload
rtcp
like ascii, but only RTCP packets
short
RTP or vat data in tabular form: [-]time ts [seq],
where a - indicates a set marker bit. The sequence number
seq is only used for RTP packets.
844525727.800600 954849217 30667
844525727.837188 954849537 30668
844525727.877249 954849857 30669
844525727.922518 954850177 30670
rtpsend
<time> RTP
v=<version>
p=<padding>
x=<extension>
m=<marker>
pt=<payload type>
ts=<time stamp>
seq=<sequence number>
ssrc=<SSRC>
cc=<CSRC count>
csrc=<CSRC>
data=<hex payload>
ext_type=<type of extension>
ext_len=<length of extension header>
ext_data=<hex extension data>
len=<packet size in bytes(including header)>
<time> RTCP (SDES v=<version>
(src=<source> cname="..." name="...")
(src=<source> ...)
)
(SR v=<version>
ssrc=<SSRC of data source>
p=<padding>
count=<number of sources>
len=<length>
ntp=<NTP timestamp>
psent=<packet sent>
osent=<octets sent>
(ssrc=<SSRC of source>
fraction=<loss fraction>
lost=<number lost>
last_seq=<last sequence number>
jit=<jitter>
lsr=<last SR received>
dlsr=<delay since last SR>
)
)
rtptrans
rtptrans
[host]/port[/ttl]
[host]/port[/ttl] [...]
Authors
Program history
Last updated
by Henning Schulzrinne