SIP Authentication methods available?

SIP Authentication methods available?

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SIP Authentication methods available? Ramon F Herrera 02-07-2007
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Posted by Ramon F Herrera on February 7, 2007, 4:51 pm

I have an AS-5300 used by several SIP phones (mostly through Asterisk)
to dial from VoIP-land to POTS-land. My current "authentication"
method is rather draconian, poor and inflexible: I just played with
the routing tables to make sure that the only phones that can have IP
connectivity with the voice gateway are those in certain subnets.

Now, I would like to move up to something that provides better
granularity. I understand that Radius is an option? There is also an
md5 digest authentication method which may or may not be supported by
the Cisco gateway?

I am guessing that another (very inefficient) solution would be to set
the 'canreinvite=no' option in Asterisk, thereby forcing it to be in
the middle of all calls. That would work, right? This solution really
sucks.

Can somebody knowledgeable discuss and/or point to some documentation
on this topic?

TIA,

-Ramon F Herrera

ps: my phones are Linksys/Sipura. Tips on configuring them for
authenticated calls are most welcome.


Posted by Hongtian on February 8, 2007, 12:02 am
Most Cisco gateway can support MD5 digest authentication. If you have
a MS-windows platform, maybe you can try miniSipServer. It is also
support this authentication and very easy to use.


On 2=D4=C28=C8=D5, =C9=CF=CE=E75=CA=B151=B7=D6, "Ramon F Herrera" <r...@con=
exus.net> wrote:
> I have an AS-5300 used by several SIP phones (mostly through Asterisk)
> to dial from VoIP-land to POTS-land. My current "authentication"
> method is rather draconian, poor and inflexible: I just played with
> the routing tables to make sure that the only phones that can have IP
> connectivity with the voice gateway are those in certain subnets.
> Now, I would like to move up to something that provides better
> granularity. I understand that Radius is an option? There is also an
> md5 digest authentication method which may or may not be supported by
> the Cisco gateway?
> I am guessing that another (very inefficient) solution would be to set
> the 'canreinvite=3Dno' option in Asterisk, thereby forcing it to be in
> the middle of all calls. That would work, right? This solution really
> sucks.
> Can somebody knowledgeable discuss and/or point to some documentation
> on this topic?
> TIA,
> -Ramon F Herrera
> ps: my phones are Linksys/Sipura. Tips on configuring them for
> authenticated calls are most welcome.



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other useful resources:
The Federal Communications Commission (FCC)
Telecommunications Industry Association
Electronic and Software Security Products and Services
International Telecommunication Union

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