Hi Dan,
SIPp is quite reliable in sending RTP, but to be sure, use
tcpdump -i ethX -s 0 -w filename.pcap host 1 or 2 or 3 ...
Replace:
"ethX" with the real name of the interface you use to send packets
towards the IVR
"filename.pcap" with any file name you find appropriate
"1 or 2 or 3 ..." with an or-separated list of IP addresses of the IVR,
some VoIP systems use several addresses for SIP and RTP
Start tcpdump before starting your scenario, and when the scenario
finishes, press Ctrl-C in the tcpdump window and then open the
"filename.pcap" in Wireshark to see what has been sent.
Possible reasons why IVR may miss some of the DTMF events:
- packet loss in the network between the SIPp machine and the IVR (this
is most likely; if you could run tcpdump simultaneously also at the IVR
side, comparison of the two files would instantly confirm or deny this
hypothesis)
- IVR confused by non-contiguous RTP sequence numbers, timestamps
changing faster or slower as compared to actual time offsets between the
packets, maybe even SSRCs caused by replaying pre-recorded pcaps (not
much likely as most digits have been detected properly but who knows
what time gaps are sufficient that the IVR would tolerate gaps and
regressions in RTP packet numbering). But be sure Wireshark _will_ be
confused at least by the seq no chaos so the automatic analysis tools
like "Analyse RTP stream" will report tons of missequenced packets.
- the machine running SIPp overloaded during sending (not much likely
but cannot be ignored). Check whether tcpdump hasn't reported dropped
packets; if it did, it is an indication that the machine may have
problems with throughput.
If you don't use Wireshark regularly, you can attach the capture file to
a reply to this, I'll have a look.
Pavel
Dne 12.7.2017 v 16:24 Dan Knopp napsal(a):
Hi,
The attached scenario is entering some DTMF via RFC 2833 to navigate
and test an IVR.
Should enter 2 then 654321# followed by 1234# and then 5855551234#
What I see reach the server is 2, 65431# then 234 then 5855551234
Did I miss something? Is the SDP I have programmed in the initial
INVITE correct?
How reliable is SIPp in sending RTP packets, are enough being
dropped/skipped or lost that I don't see some of the DTMF digits reach
the server under test?
Version information:
[dknopp@dknopp-workstation workspace]$ sipp -v
SIPp v3.5.1-TLS-SCTP-PCAP-RTPSTREAM built Mar 8 2017, 16:46:11.
Thanks,
Dan
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