[asterisk-users] SRTP enabling
Abdul
abdul_zu at yahoo.com
Sun Jul 16 21:45:34 MST 2006
Hello,
In some countries i found that they are blocking SIP port 5060
so instead of this i change to another port 1221, and its work
well. But in one country the are not blocking SIP but they are
playing with RTP packets, if they filtered it is VoIP RTP they
are doing something called party cannot hear or some time caller
cannot hear but called party can hear well.
So i cosider to use SRTP to make encryption. and i am using
my asterisk in VPS so i have full control to manage the server.
If you guys have better Idea to prevent such kind of issue, it
will be good for us.
Abdul
Most of the blocking in other countries, was not for RTP traffic, but for signaling traffic (SIP usually, Mexico x Vonage comes to mind).
You are sure they are blocking RTP traffic ?
And, from what I understand, in some places the gov. forced the ISPs to remove the blocking (at least, I heard of one such a case in Brazil, a DSL provider started to block SIP, and Anatel, Brazil gov. entity that regulate telephony and others, asked them to remove the blocking, others with more knowledge of the case may be able to add their remarks) Blocking SIP if you control the server is somewhat easy to prevent (if is a plain dumb UDP port 5060 filtering), just have your server listen in another UDP port...
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