atcomsystems.ca/forum Forums VOIP Phones & IP PBX VoIP & Cloud Phones, Hosted PBX, General IP Phones SIP Registration

I'm looking to try to find a difinitive answer in regard to registration of SIP end points on a PBX (Comdial MP5000) so that SIP to SIP calls can be made from one IP end points in one MP5000 to SIP end points in another MP5000 or from an MP5000 SIP end point to another manufacturer's SIP end point. I am not talking about the ability to call from one location to another. I am talking about true SIP calling. As I understand it there is no public registrar of SIP addresses like there are for email and domain names. Comdial tells me that at present a SIP call can only take place to another SIP device within the same switch. This allows the use of SIP features such as video soft phone and presence management and viewing. SO if I have three MP5000 systems, I have to register the SIP soft phones all in one switch (networked swtitches) to be able to do these things. Soon they say that this issue will be resolved so that any SIP end point on the network can register in it's own switch and all features will be available across the Comdial MP5000 network. Eventually there is supposed to be some sort of public registry server or servers, or at least a way for Comdial to register end points from another manufacturer's switch so SIP to SIP calling can take place.

For anyone who reads this that doesn't know exactly what I am talking about, You know, when you send an email to [email protected], he does not have to be residing (registered) in the same server you are in to get his email. He can be anywhere in the world. But with SIP he does have to reside in your server (registered SIP end point). I am not talking about Vonage or any of the other SIP IP phone companies. I'm interested in the advancement taking place within SIP based IP PBX systems. And I am using Comdial MP5000 only as an example. If someone else's system works differently, I would like to know about it.

Here are the implications of TRUE SIP to SIP calling as I understand the promise of it: I can use my SIP based end point (hard phone or soft phone) to call any SIP end point anywhere in the world and never really be concerned with the call going over the public telephone network. If I am calling from my office to Germany and the address of the SIP end point in Germany is [email protected], then my system looks to a registry server to identify where [email protected] is located and checks to see if he is available to take my call and if he is, we're connected. (I know there is more to the steps involved so don't pick that part to death). If I'm using a video soft phone and [email protected] is not, the call still goes through, I just do see him. In some applications the ability to do SIP to SIP calling would eliminate the need to set up networked switches, for instance if all the customer was looking for was a way to save on toll charges and call other individuals direct, this would work great. We have a need for this application where there is going to be multiple MP5000 switches throught the US and a number of Alcatel OmniPBX's in Europe. I'm out of breath so tell me what you think.

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Hopefully this will answer your question, but this is something that Allworx can do for you.

Network up to 99 Allworx PBXs for site-to-site direct dialing without the need for a central Proxy/Registrar. Each PBX acts as a standalone phone system for its local users and provides the capability to dial to another PBX over the Internet using the integrated dialing plan. Site-to-site calls are free because they are passed from one PBX to the other over the Internet.

Interface with up to 100 third party gateways or proxies for the purposes of routing calls to non-Allworx PBXs. SIP-capable PBXs as well as legacy PBXs fronted by a SIP gateway can call (also be called by) by an Allworx user across the Internet without toll charges. The dialplan for each SIP endpoint directs where the calls should be routed over the Internet.

Allworx also provides a capability to bypass Internet access outages by routing calls over traditional PSTN services if available. If you have any further questions, give us a call and we can doa conference call with you to answer your questions more in depth.

Joshua

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https://www.allworx.com

All In One Solution For Small Business"

Great. Now tell me how you manufacturing dudes are addressing collectively a public registration solution.

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Well that tells us all about how allworx does it but that wasn't his question.

You are confusing SIP and email/voicemail. I think that's as simple as it needs to be unless I am missing something here?

No. I've been looking into this for a while. SIP end point registration has to take place in a registry server somewhere so other SIP end points can find other end points when they try to make a call (SIP TO SIP). At present, from what I understand, there is no public registry server for SIP. With Vonage and other IP phone companies, the ATA device registers to the Vonage registry server, but the end user never knows it so the issue of registration might be a relatively unknown issue for some. The reason this registry issue is important to me is that I have a customer with multiple sites in the US and multiple sites in Europe. SIP stations in the US can't call SIP stations in Europe through IP direct SIP to SIP because they have no common place to register their addresses. In essence, the SIP station in the US registers to the MP5000 it resides in and has no idea the ones in Europe even exist. So, if you relate this to the analogy of sending an email, imagine that you could only send emails to people in your office because there was no public registry service to keep track of where other email users and domain names were located. This is where we need a manufacturer to get in and answer the question from a non-sales standpoint and from a generic product standpoint.

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"You are confusing SIP and email/voicemail. I think that's as simple as it needs to be unless I am missing something here?"

Now there are going to be people who will say I can call from my SIP phone anywhere in the world. True, but that is not what I'm talking about. There are also people who will say you can install an IP network. No I can't because that QSIG crap has not been worked out properly, especially when you have a switch in another country, specifially Europe (Germany). You can assume " this guy is confusing email/voice mail" and I would agree if I had not put so much time into this already. But the fact of the matter is that to make a true SIP to SIP call without going through the PSTN or through some kind of gateway, you have to be able to find each other. When this is worked out, telecommunications as we know it will be turned upside down.

ALLWORX: Get Me An Engineer For This One And Post Here So I Can Proove I Am Not Crazy.


[This message has been edited by MARK3906 (edited April 08, 2005).]

Do you know what SIP is? SIP is simply a loose standard. SIP phones/software have to authenticate to a server usually through MAC adressing. I think you are WAY confused here. What you want is something like netmeeting ala .323 gateway which already exists. There are also comanies in which you have to pay for non IP calls you make on the IP service. email and voicemail have NOTHING to do with SIP protocal. QSIG works great for IP non-ip networking of switches. SIP phones have to authenticate to a server verifying a valid MAC address. Email/domain servers do not require this and work totally different that VOIP or in this case TOIP tech. You are comparing apples and oranges. Until SIP is standardized and works properly what you want is years away if it's ever going to be possible.

Here is a good laymans link for you to better understand.

https://www.cs.columbia.edu/sip/drafts.html


[This message has been edited by Coral Tech (edited April 08, 2005).]

[This message has been edited by Coral Tech (edited April 08, 2005).]

I don't mean for this to drag out but I have the issue of SIP registration dead in front of me. As loose at the standard may be, I have an issue to deal with and have my answers from Comdial as to where they are going with it. What I hoped to get when I posted this was a manufacturer's position. You're input is great and maybe I don't express myself properly, but I know what I need to know. I just have not been connected to the proper people to get the info I need. SIP has issues. I know that. SO did voice mail when it first came out. So did touch tone phones, for that matter. You're a big fellow doing large switches. I respect and admire that. If I need to rephrase my post, please educate me. That's why I'm here.

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No problem, I didn't mean to offend. SIP is still young, where it is going who can say. For VOIP it's going to be huge. I mean, you are going to be able and go to walmart and get a $50 SIP phone and attach it to your phone system. Right now the dedicated proprietary VOIP phones are expensive....this is going to change and change big in the next year or so.

My Comdial EP300's already sell for the same price as my 24 button digital sets. They look like the same phones. One is digital, one is SIP. For me the future is here. and check this out. MP1000 with 8 CO line capacity and up to 100 SIP end points (40 of them EP300 hard phones). It has issues but it also has limited UM, voice mail built in, desk top phone admin for the user. The product is in my hands right now. I know it's not ready to sell and may be a year or so away from being usable. But I already know what it cost me in hardware ( which I know is not the only cost associtated with it) but ask any other Comdial guy who can get it (gold dealers) and they will tell you, the price is unbelievable. Remember when you first saw a DVD player for $89.00 and you thought, how can they do that. Now you can get one for $29.00. Same thing with VOIP gateways. I looked into Net2Phone. Gateway was $700.00 two years ago. Buy one and save some on my long distance, right? Then within a year they were $189.00. Now you can get them for free if you're willing to go on the internet and sign up. The soup is going to get thick with this stuff and it's not H.323 that's going to be the issue for small systems.

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