If you tried SIPp you would know. The idea is the same, but this one is much
easier to use and will eventually be able to do most of what SIPp can.
> How it different from SIPp? http://sipp.sourceforge.net/
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Sip
You can create a complex scenario and therefore automate testing. At the moment
it is not supported to:
a) have multiple calls at the same time
b) load scenario, run it, close scenario, loop through this procedure for
different scenarios.
Documentation is at: http://sites.google.com/site/sipinsp
scenarios. Just
released version 0.7 represents a significant milestone. The new release can
send RTP packets from a captured pcap file.
While on its way to maturity, please give it a try.
Best regards,
Zarko Coklin
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Richard, if your device is UA it has to create INVITE request and send it to
OB. UA does not add OB into routes. OB will have to add itself into routes. It
knows what type of routing it supports and will add appropriate route type.
>But how about for initial INVITE? Before adding the OB to rout
You are talking about reSIProcate: http://www.resiprocate.org
BTW, if your contribution will be valuable to others you can consider donating
the code.
Boardwalk for $500? In 2007? Ha! Play Monopoly Here
ement. Whether you follow 3a or 3b depends what is
found in top-most route. Since strict routes are still in circulation UA should
be able to support both logics.
Regards,
Zarko Coklin
Ready
for the ed
Thanks to all who responded!
I find P-Asserted-Identity to be the best solution and here is why.
1. Because of its simplicity it is very likely already supported by major
telecom companies.
2. What Michael suggested may be used but has serious limitations as pointed
out by Chandra. In addition,
In some cases (for example, when phones are connected to IP-PBX) it is required
to change called party's display name. How is this done?
I tried following on couple of different IP phones but it did not work :-(
IP Phone IP-PBX
--INVITE (sip:[EMAIL PR
