11 Mar
2004
11 Mar
'04
8:18 a.m.
On Mar 11, 2004 at 09:05, Nils Ohlmeier <nils(a)iptel.org> wrote:
Hello,
On Thursday 11 March 2004 08:53, Alex Bligh wrote:
Depends on the codec and network you use. 10000 gsm would be easy. You
also don't need to encode some valid stuff, you generate on packet and
just slightly modify it before sending.
On gigabit you can send 120Mb/s over udp (athlon 2000+).
Andrei
Before I reinvent the wheel and go write it, is
there a utility which will
simulate making & receiving calls (i.e. SIP conversations, **plus RTP**) in
scale (to answer a question like "will my SIP+RTPproxy deployment scale for
{100,1000,10000,100000} users using the following parameters for average
call duration, % phones in use etc."
there are definitely call simulators with audio/RTP simulation available on
the market. At least they were present at the last SIPit.
If not, anyone have any recommendations for a
(free) stack that does a
decent UAC/UAS job, including audio, and will run without GUI?
The tools mentioned above are for sure not freely available.
And if you plan to develope such a tool: keep in mind what will reach its
bottleneck first the tester or the tested device. RTP simulation with the
numbers you are looking for is probably only possible with specialist
hardware and not with normal PC (at least these guys at SIPit had special
hardware boxes for these tests and i guess they know why). I guess that a
normal PC will reach its own limits at 1000 or at least at 10000 simultaneous
RTP calls, maybe even earlier.