Because RTP ports are assigned dynamically (and not necessarily symmetrically) during call setup using SIP, you need a SIP aware firewall. Without one, you may get SIP registration, but usually one-way/no audio (RTP).
Most hotels and hotspots do NOT support SIP - either because they run cheap firewalls/routers or because VoIP competes with other services they offer. IAX is single port and symmetrical so even cheap firewalls/routers can pass this without additional setup. Our consultants travel across North America and we finally gave up on SIP phones because of these hassles. (Trying to explain SIP, NAT, IP Masquerading, symmetry, RTP, etc to tech support for each hotel was a time waster. *We* know what the NAT/IP Masquerading issue is - but that doesn't help some tech support guy in India assisting Marriott customers). Perhaps wherever you are located the state of firewalls/routers is different. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Friday, October 30, 2009 8:37 AM To: Asterisk Users List Subject: Re: [asterisk-users] [IAX] Recommended soft- and hardphones? My experience does not support your conclusions. In my personal observations of situations in which I have been involved, most allegations of serious SIP problems related to source NAT ("IP masquerading") are exaggerations stemming from lack of subject matter comprehension. This is bearing in mind, duly, that SIP and NAT *is*, inherently, a problematic equation - of that, there can be no question. But I've never had problems getting a SIP soft phone to make and receive calls from anywhere I've taken it. The only substantial problem I've run into is that many NAT gateways lose UDP state quite quickly, so after a certain period of inactivity, calls cannot be received; this is solved by decreasing the re-registration interval, or increasing the frequency of state-sustaining SIP OPTIONS pings, etc. Many service providers have implemented such steps since the last time I was involved with this problem seriously. I'll take your word for the fact that IAX may be easier, though. Michelle Dupuis wrote: > I assume you're kidding?! > > RTP is mangled/blocked by most hotspots and mid-size company firewalls... > > IAX is often the only way our staff can connect while on the road. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex > Balashov > Sent: Friday, October 30, 2009 8:03 AM > To: Asterisk Users List > Subject: Re: [asterisk-users] [IAX] Recommended soft- and hardphones? > > Vincent wrote: > >> Since SIP/RTP is a pain to use with road warriors who need to connect >> from any location over the Internet, I'd like to get them some IAX >> phones instead. > > What gives you that idea? > > -- > Alex Balashov - Principal > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users