Knowing the status of a registration attempt would be useful. One would not necessarily remove the local registration if the external attempt failed. Local calling would still be possible. As well a remote family member for example could take advantage of such dual registration and could take calls that came through a service provider or calls that originated on the LAN.
Using dynamic DNS acquaintances could also call into the LAN without going through a service provider. The UA only concerns itself with the username not the domain. Using aliases and fixed IP addresses on the LAN you can even ring individual phones. I use my proxy much as I have described. I am currently back and forth between two residences. Using an SPA3000 and my proxy, my PSTN calls ring at both places. As well my wife and I call each other without going through the PSTN or a voip service provider although the service provider handles the voicemail. I see myself as a SIP promoter. A good way for a beginner to get experience is to set up a home or small office system. As I said before I understand that Opensips must cater to the service provider but nurturing the newbies is good. They are the future. On Friday 22 May 2009, Bogdan-Andrei Iancu wrote: > Hi Robert, > > Interesting points of view :). > > But there is a huge difference between a proxy acting as a mid-registrar > (with no auth knowledge, no user knowledge, but simply following the > master registrar decisions in a blind way) and proxy doing actually > account and registrar management (like you described in the scenario > with hiding registrations). > > I'm not saying is not possible, but the question is where a proxy stops > and where an ALG starts :). > > Anyhow, I will add on the TODO list the possibility to do a register at > reply time, so that we can use opensips as mid-registrar. > > Regards, > Bogdan > > Robert Dyck wrote: > > I like the idea that we could maintain a local registrar that accurately > > reflects the remote registration. I would go so far as to say that it > > would be useful to be able to optimize opensips as an ALG. Some people > > could use a proxy as an edge device on a LAN. You could have several > > phones with the same user name register with a service provider. Some > > service providers limit the number of contacts per AOR. This would have > > the side effect of keeping the signaling associated with forking on the > > LAN. By detecting local calls, media could also be kept on the LAN. Such > > an edge device might have a dynamic IP address. It would be helpful if > > opensips could conveniently be setup to fix the contacts. Using the > > received address does not work when the phones are on the same LAN as the > > proxy. > > > > Opensips has a bias toward service providers. This is quite > > understandable. However I feel with a bit of tweaking it could serve as > > an ALG for the home owner or a small business. > > > > On Wednesday 13 May 2009, Bogdan-Andrei Iancu wrote: > >> Hi Jeff, > >> > >> theoretically yes, because you have all the needed information > >> (hmm...maybe except the NAT status from request, but you can store it > >> via transaction)....Practically, the save() function does expect to > >> receive a request only, so it must be changed to work with a reply also. > >> > >> Regards, > >> Bogdan > >> > >> Jeff Pyle wrote: > >>> I've thought a lot about this as well, although I haven't taken it > >>> nearly as far as John has. > >>> > >>> A thought: is it possible to do a save() in the reply route, only upon > >>> a 200 OK from the end registrar? > >>> > >>> > >>> - Jeff > >>> > >>> On 5/12/09 5:09 AM, "Bogdan-Andrei Iancu" <bog...@voice-system.ro> wrote: > >>>> Hi John, > >>>> > >>>> This mid-registrar approach may work but it is not 100% correct as > >>>> OpenSIPS (as mid-registrar) does not obey the actions of the final > >>>> registrar (Asterisk). Ex: > >>>> - Asterisk may forbid the registration and you already saved the > >>>> registration on OpenSIPS > >>>> - Asterisk may change the Expire time while to saved the > >>>> registration with the expire sent by client. > >>>> > >>>> Anyhow, ignoring this aspects, lets go further :) : > >>>> > >>>> 1) is the registration scenario working ok? if not what is the exact > >>>> problem (some trace will help). > >>>> > >>>> I will wait for you answer before moving further with the calling > >>>> stuff. > >>>> > >>>> Regards, > >>>> Bogdan > >>>> > >>>> John Morris wrote: > >>>>> After several days of playing with OpenSIPS 1.5.0 and RTPProxy 1.2.0, > >>>>> I have a partially working SIP+RTP ALG configuration, and have gotten > >>>>> stuck. I could use some general advice from the list. > >>>>> > >>>>> The company has an Asterisk/FreePBX server on an internal network, > >>>>> and the CEO wants to use a SIP phone from outside. Because the sip > >>>>> alg iptables module isn't working, and in preparation for another > >>>>> project, I started investigating OpenSIPS for use as a border proxy > >>>>> to connect phones across NAT (and, the next project, to route a SIP > >>>>> trunk over a VPN from the network of a DSL+phone company that > >>>>> intermittently blocks SIP traffic in hopes of plugging revenue > >>>>> leaks). > >>>>> > >>>>> The network looks like this: > >>>>> > >>>>> SIP UA <-> home NAT gateway <-> Internet <-> OpenSIPS server/NAT > >>>>> router <-> Asterisk > >>>>> > >>>>> The standard opensips.cfg file doesn't work as is. The SIP phone > >>>>> needs to register to the Asterisk server directly. In addition, it > >>>>> seems there is extra logic needed to support multiple network > >>>>> interfaces (mhomed=1 only partially solves the problem). > >>>>> > >>>>> The way I've gone with this in testing is to relay REGISTERs to > >>>>> Asterisk, but after a save("location","0x02") to enable a > >>>>> lookup("location") on messages originating from the PBX. The phone > >>>>> is configured with an outbound proxy, and all packets to the proxy > >>>>> matching "uri==myself" are thrown away. This worked great on the > >>>>> single-interfaced, internal test installation. Now that there are > >>>>> multiple interfaces involved, things are breaking again; ACKs and > >>>>> BYEs are sent out the wrong interface, and RTPProxy is behaving > >>>>> strangely in bridged mode. > >>>>> > >>>>> There seem to be no good configuration examples for either > >>>>> multi-homed proxies or for proxies that relay REGISTERs. This makes > >>>>> me think that I'm going about this the wrong way. > >>>>> > >>>>> Also, I have looked at other software, like siproxd, opensbc and uh, > >>>>> that other b2bua that functions as an SBC, but none of those seem to > >>>>> allow this REGISTER pass-through function. > >>>>> > >>>>> What is the best approach for this scenario? The above approach of > >>>>> relaying REGISTERs to Asterisk? Is there maybe another approach > >>>>> where phones register to OpenSIPS directly, and OpenSIPS in turn > >>>>> somehow sends another REGISTER to Asterisk? Or am I missing the idea > >>>>> completely? > >>>>> > >>>>> I'd appreciate general pointers about how to proceed. I've been > >>>>> putting some Asterisk and FreePBX tutorials and CentOS RPMs on > >>>>> http://www.zultron.com, mostly aimed at small office-like > >>>>> environments. Looking through various lists, this seems a highly > >>>>> sought-after configuration. If I succeed, I'll document it in hopes > >>>>> of filling the gap in this sort of example. > >>>>> > >>>>> Thanks > >>>>> > >>>>> John > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users@lists.opensips.org > >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >>>> > >>>> _______________________________________________ > >>>> Users mailing list > >>>> Users@lists.opensips.org > >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> _______________________________________________ > >> Users mailing list > >> Users@lists.opensips.org > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users