Not sure what exactly you are asking.
You can use play_pcap_audio to replay a single RTP stream. The .pcap
file to be played must not contain anything else but that stream. SIPp
then changes the source and destination addresses and ports according to
actual values used locally (from auto-media-port in the scenario and
from the mi command line parameter) but keeps the rest (ssrc, sequence
numbers) unchanged.
So to replicate a failed call scenario based on a pcap, you have to
- use a display filter which shows only that single RTP stream and use
"export selected packets -> displayed" to save them into a new .pcap
file which you would then use for play_pcap_audio,
- use a display filter like "sip.Call-ID==your_call-id" to show only
messages belonging to the SIP dialog in question and use "export
selected packets -> displayed" to save them into another new .pcap(ng) file
- disable the SIP dissector, set Wireshark to display non-dissected data
as text and run tshark from command line:
tshark.exe -r your_single_SIP_dialog_file_name.pcapng -T fields -e
data.text > your_file_name.txt
to extract the SIP conversation from the pcap(ng) file into a text form
(switching on "display SIP text" in SIP preferences doesn't currently
work well with tshark)
- edit your_file_name.txt to add the XML tags around the messages to
convert it into a SIPp scenario file
Pavel
Dne 11.4.2018 v 13:59 Franklin Angulo napsal(a):
Hi. I am using SIPp to create test scenarios that simulate real SIP
communication scenarios between server and client that have generated
errors when installing a real VoIP system. I have a capture .pcapng of
Wireshark and I want that based on this capture I can generate a SIP
scene with which I can perform my tests in the laboratory with my SIP
phone or the server depending on the case that occurs. What I do first
is convert the .pcapng to .pcap format to include the .pcap in the
statement <exec play_pcap_audio = "pcap / File name.pcap" /> . My
scenario is:
IP Phone IP Server
IP Phone Invite ---> IP Server
IP Phone <- 100 IP Server
IP Phone <- 407 IP Server
IP Phone ACK ---> IP Server
IP Phone Invite ---> IP Server
IP Phone <- 100 IP Server
IP Phone <- 180 IP Server
IP Phone <------------------------------------ RTP (G711U)
IP Phone <- 200 OK SDP G711U IP Server
IP Phone <- 180 IP Server
IP Phone Cancel ---> IP Server
IP Phone <- 481 Call IP Server
IP Phone ACK ---> IP Server
IP Phone <- 200 OK SDP G711U IP Server
IP Phone <- By IP Server
Is it possible to generate this scenario with the audio and the SIP /
SDP that the .pcapng file from Wireshark gives me, having compiled the
SIPp with ./configure --with-pcap or does it have to add another module?
Thank you very much for the information!
Carefully greetings,
Franklin
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