[Sipp-users] Pcap_play and B2BUA (Like Asterisk)

[Raka <wir...@ya...>]

Hi,

Has anyone here tried the pcap_play with features in
conjuction with Asterisk? 

The scenario: SIPp "A" dials an extension in Asterisk,
and Asterisk will connect it to SIPp "B". Then, SIPp A
will send RTP packets that will be forwarded by
Asterisk to SIPp "B"). Please notice: I set
canreinvite=no in the /etc/asterisk/sip.conf

I've never been able to successfuly run my test using
pcap_play. The problem was primarily because we can
not know beforehand the destination port of the RTP
packets (because it's randomly picked by Asterisk for
each new call). So recording the packets using
ethereal when talking using a SIP softphone and then
using that recorded packets from SIPp wouldn't work,
because the dest. has changed in the new call.

Please correct me if I said something stupid in my
analysis above ;) -- because I'm not sure if I
understand the situation accurately.

Best regards,
Raka -- http://www.jroller.com/page/donraka

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