Sip client on hot spot


is it possible to connect a sip user from remote into my asterisk server in main office. im using n80i with no problem in main office through wifi
now when im out from office and stay in an internet hot spot id like my n80i to connect to my * at office so i could make out going calls or receive incoming calls

here is my ideal schematic diagram

n80i wifi conected (www-hotsopt/dynamic ip)<----->main office where * is behind firewall with static ip

what would be the pre requisite?

Many thanks

Well, so long as you have a DNS entry as your SIP proxy (asterisk) that resolves properly internally and on the internet, you should be able to open the SIP and RTP ports on your border router. Keep in mind that there are security implications to this… so I would highly suggest a good router that can do stateful inspection on SIP and RTP packets. And, of course this goes without saying, securing your asterisk box as best as possible. :smile:

Good luck. -Cheers, Peter.

how about in my n80i side? dont i need to fix anything? as it is connected
to dsl (with dynamic ip) and then internet cloud and then to my firewall in asterisk side.
what if someone dial my local number from LAN (office),shall asterisk could make an outbound route going to that hot spot router?

i can see a possible route coming in to asterisk but i dont think in going out from * to my n80i

please help give some options

Well, by opening your SIP and RTP ports on the internet, your phone will use DNS to resolve your public IP when at a hotspot and connect to your open SIP and RTP ports.

Dynamic IP will make things a bit more troublesome. But there are some dynamic dns services available that you could try. - never used them myself.

The open ports will allow your phone to connect into and register with asterisk with your current hot spot’s public IP address.

The main question that is unknown is if the hotspot’s router understands SIP and RTP to forward return packets inward. It really depends on the hotspot…

-Cheers, Peter.

ok i figure it out, ok since we are here and mention about SIP and RTP ports could you tell me please what is the best port for RTP. i have read from AsteriskFOT that it ranges from 10000-20000 however in your working experiment/eperience what is the exact port value?

many thanks

There is no “exact port value”. It is negotiated as part of call set up each time. The phone usually picks its own port and the server give out an open port based on what is available and idle. 10K to 20K is the normal range, but some go outside that as well.

much clearer than yesterday(RTP ports)

many thanks


after reading this topic a second scenario came into my mind, what would you do?

In my office there are two people on any place around the globe. Both are using a WiFi phone and are logged on my asterisk server at the office.

Now is it possible that my * Server only does the connection between both and all the RTP traffic will go only between them?
Do I need NAT for that?

Maybe the server is located in Rome and both WiFi peers are in australia. Interconnecting between the peers only would better the connection, wouldn’t it?

kind regards

as what i have understand RTP is being and will be used by the request of SIP client. so RTP must be in * server LAN.

RTP is the media of conversation and in order for two SIP client to talk with respect to *, RTP must be register and monitor by * serber.

not so sure but i have glimpse with this scenario

thank you for your reply,

but setting up the phones canreinvite=yes it should be possible to route the RTP traffic from peer to peer after this “reinvitation”.
As I read it should be possible in a NATed Layer 2 network to do that if my personal IP adresses are being routed and the ports are not firewalled.

I’ll do a chart/picture of that at home and will forward it here for further discussion.

kind regards