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NAMErtpsend —
generate RTP packets from textual description
SYNOPSIS
DESCRIPTIONrtpsend sends a stream of RTP and RTCP packets with
configurable parameters. The packets are sent to the given
address/port, optionally with a
time-to-live value of ttl. This is intended to test the
RTP features of the receiving end.
By default, Within the input, each entry starts with a time value, in seconds, relative to the beginning. 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. RTP and RTCP entries look like this: <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> ) ) The options are as follows:
SEE ALSOrtpdump(1), rtpplay(1)AUTHORSrtpsend was written by Henning
Schulzrinne
<hgs@cs.columbia.edu>,
with enhancements by Ping Pan and
Akira Tsukamoto
<akira.tsukamoto@gmail.com>.
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