On the contrary, SIP-aware ALGs mostly do more harm than good. While their purpose is noble, their implementation is frequently lacking/ incomplete and conflicts with existing far-end NAT traversal approaches taken by most service providers these days. These are also present in commercial PBX equipment, e.g. along the lines of nat=yes in sip.conf.
Almost all UAs symmetrically signal and the overwhelming majority do symmetrical RTP as well. This used to be more of a problem a few years ago. I don't disagree that IAX may be easier still. I just haven't found that the alleged problems of SIP and NAT live up to the hype now that almost every commercial service edge does far-end NAT detection and draft-comedia style RTP handling. -- Sent from mobile device On Oct 30, 2009, at 8:58 AM, Michelle Dupuis <[email protected]> wrote: > 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: [email protected] > [mailto:[email protected]] 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: [email protected] >> [mailto:[email protected]] 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 _______________________________________________ -- 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
