[Asterisk-Users] Budgetone 101 phones ? SIP through NAT ?

[Miguel Ruiz Velasco Sobrino]

>I'm new to VOIP.  We are thinking of setting up a VOIP system between a
>couple remote offices.  I've been lurking on this group for a while. 
>
>What is the consensus on these phones:
>
>http://www.netvoice.ca/grandstream/budgetone101.htm
>
>
>I'm confused about the SIP protocol... can a SIP phone be located behind
>a NATing firewall ?
>When people use asterisk on a broadband connection used for data and
>VOIP, do they put the asterisk machine behind a firewall or do they put
>the firewall on the asterisk machine ?

You will have a rather big problem doing that, you will likely end in the one-way audio
scenario; to overcome that you may use STUN or TURN or an application level gateway
(example: *). STUN is rather complicated to configure and I advise that only if you
REALLY need it; never used TURN, but is almost the same. Now, you can put a multihomed
machine (with an IP and posibly a NIC in each side of de fw), to do the passing of the
calls to an external provider.

But if you will link different offices each one with an * server, use IAX2 in the middle,
it doesn't suffer the on-way-audio-problem, it's NAT friendly (if you have one or many in
the middle), and if you enable trunking, saves a good bunch of bandwidth for each
additional conversation.

>Is anyone using QOS throttling when sharing VOIP and data on the same
>broadband connection ?  Is it necessary ?  

I use QOS to prevent de data traffic eating all the available bandwidth and render VoIP
unusable. You can use a HTB with SFQ in each bucket, or a pfifo, or if you are brave and
know what are about to do, use diffserv.

>Thanks.
Miguel Ruiz Velasco

=====
Miguel Ruiz Velasco

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