Hi All,
Hopefully the following issues are already tackled by more experienced
people or I have been going for uncommon cases. Any help is appreciated:
- SIPp (3.3, openssl 1.0.1e) to Kamailio (4.2.1, openssl 1.0.1e) TLS
session lasts only 3 seconds. Kamailio says it read "EOF" and breaks
connection. Anybody experienced problems with TLS? PJSIP & Linphone seems
to work fine with Kamailio over TLS.
- auto_media_port is only available for "playing side". For "echo" port,
you have to guess available ports so that you can pass it to "-mp". If you
attempted to use an already "in use" port you are out of luck. SIPP will
not attempt to find any available ports. Any workarounds?
- I don't care about "remote control port". As soon as I run more than
100 instances on the same machine (50 UAC, 50 UAS) it starts colliding just
like the "-mp" port issue. Any workarounds?
- I will be sending parallel RTP streams. How can I verify if all
streams were sent/received properly? How can I find out about packet loss
etc?
- SIPp, can be either a UAC or UAS. This makes a scenario like the
following impossible:
- perform REGISTER (transport=TCP)
- expect INVITE
The only workaround was to forget about TCP:
- I performed registration using a simple digest based scenario,
recorded discovered "received" and "rport" parameters along with contact
header and exit SIPP
- Run another simple UAS scenario binding to the same port from where
registration was performed & re-used contact header previously recorded.
It seems like this will continue to be impossible in a single SIPP until
polycom multi-scenario capabilities are ported back.
Best,
Onur.
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