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w1ve
Premium Member
join:2007-12-28
Hancock, NH

w1ve to GraysonPeddi

Premium Member

to GraysonPeddi

Re: will SIP URI's calling ever take off

Don't use ZoneEdit. try Editdns.net. Free. And they support SRV records.
w1ve

w1ve to GraysonPeddi

Premium Member

to GraysonPeddi
said by GraysonPeddi:

It'd be nice if an SIP URI can be the same as an e-mail address like this:

e-mail: yourname@yourdomain.com
sip:yourname@yourdomain.com

But then you'll have to register a domain name (which I did and I have my own) and find an SIP provider that can allow you to use your domain name and map a username to it.
It seems you do not understand SRV records.

Go buy a cheap domain, like ilikevoip.info, for $1 a year.
Use Editdns.net for your DNS. (There are lots of free or nearly free DNS providers that will let you have SRV records)

Let's say you have a dynamic IP on myisp.com, but you use
a free Dynamic dns account to give your dynamic ip a dns name like "myata.ilikevoip.info".

All you do is create a SRV record on your ilikevoip.info domain that points to myata.likevoip.info -- boom! SRV dialing works:

email dude@ilikevoip.info and sip uri dude@ilikevoip.info -- all working, nothing special required -- and about 15 minutes of time. Cost: $1 for the info domain. YMMV

BTW, you can also add a jabber SRV record, then run a jabber server from your pc... then you can have your friends enter your email address as a jabber account on Google Talk, etc.

Gerry
GraysonPeddi
Grayson Peddie
join:2010-06-28
Tallahassee, FL
Ubiquiti EdgeRouter PoE
Ubiquiti UniFi AP-AC

3 edits

GraysonPeddi to w1ve

Member

to w1ve
Zoneedit is free but when I click "Start Trial" in EditDNS, that tells me that it is a paid service, but I'll try it anyway.

Though I've got to wonder if EditDNS allows for an IP change via a wget script or if there's a program for Linux (via command shell) that will do that.

Update: I already bought my domain name about two years ago when I signed up for ZoneEdit, but I'm in the midst of changing to GoDaddy (used only as a registrar service).

I am having trouble selecting proper keywords in Google Search. Can you please point me to a good free DNS hosting service that allow for adding SRV records?
w1ve
Premium Member
join:2007-12-28
Hancock, NH

w1ve

Premium Member

If Godaddy is your registrar, DNS is free, and they support SRV records.
nickdigger
join:2009-06-13

nickdigger to GraysonPeddi

Member

to GraysonPeddi
said by GraysonPeddi:

It'd be nice if an SIP URI can be the same as an e-mail address like this:

e-mail: yourname@yourdomain.com
sip:yourname@yourdomain.com
Yes, that would be great. I'd love to get a simul-ring to my phone with every "v1agra c1al1s 4 cheap" and "please your russian bride long time now" email that hits my inbox.
dalrun
join:2008-01-09
Bellingham, WA

dalrun to tom thomas

Member

to tom thomas
said by tom thomas :

you can create SRV records to send call directly to your own public IP
If your domain points to a webhost and you want to route SIP calls to your public IP address, your SRV record needs to point to your public hostname (could be your ISP hostname if 'host [your IP address]' returns a hostname).

If your domain points to your public IP address, you don't need a SRV record.

To route to a VOIP provider, I'd think that the provider would need to be able/willing to translate foo@domain to user@provider. It looks like you could route to OnSip if your domain and username are the same (»www.onsip.com/features/s ··· -hosting).

Whereforeart
@teksavvy.com

Whereforeart to tom thomas

Anon

to tom thomas
I'm intrigued by this $1 sip call business but you lose me after get a domain like iamthick.com is a Noddy explanation even possible or am I too far down the evolutionary chain?

Regarding sip mass market you need to turn all those numbers into a word like 188 gofedex no hang on it's 1800 fedexnow or is it....
And then you need it to work from all email programs just like IM
w1ve
Premium Member
join:2007-12-28
Hancock, NH

w1ve

Premium Member

You don't have to be a programmer to understand DNS. By doing a little reading, you should be able to figure it out and make it work.

I'll try to explain the basic mechanics of SRV records in the context of using them for SIP -- and I'll point to some resources that explain in more detail.

First off, If you do not own a domain, all bets are off -- because you have to have control over the DNS records. DNS records essentially map IP Addresses to domain names.

If you need a domain, find a domain name provider you like. A bonus might be that the provider also hosts DNS that supports an SRV record (GoDaddy, etc). Additionally, there are special DNS records which map specific internet services to servers. A well-known example of this is the MX record, which specifies the mail server for a given domain name. an example might be: (I'm showing the data -- not the exact format -- it varies by platform)

www.myweb.com 1.1.1.1
MX 10 mail.isp.com
MX 20 mail2.isp.com

What this says is the public IP for my web site is 1.1.1.1, however, is someone sends mail to whomever@myweb.com, mail first tries the server at mail.isp.com (whatever that IP is), and then tries mail2.isp.com if the 1st one fails.

SRV are service records. By convention, they have a specific format. For SIP URI dialing, you do not need an SRV record IF your SIP-based gear is running on the same IP as your domain name (pretty uncommon). So, if your SIP gear is on some other IP, just like MX records for mail, we want to route to your SIP gear based on what the SRV record says.

Here are a couple of good links to SRV record tutorials.
»anders.com/cms/263
»support.easydns.com/srvrec.php

In the end, you have an SRV record that says "if someone is requesting UDP SIP service from domain XXX.tld, here is the server that handles that."

The nice thing is that the server that handles the request is specified as a domain, not an IP. So, those who do not have a fixed IP can use a DNS service that works with dynamic IPs.
Your SRV record can then point to the DNS name which is updated using Dynamic DNS. (Google dynamic dns if you don't understand that part).

So, once you are set up, is SIP URI calling automatic? Well, not always. It depends on if the calling endpoint does SRV lookup.

Counterpath x-lite supports it. (Even with your browser) enter sip://joe@somedomain.com and x-lite will attempt to dial) (Note that if x-lite is connected to asterisk, you'll have to make that work in your dial plan.) Most Linksys ATAs work with SRV records.

SRV records and URI dialing open up a lot more functionality for VOIP -- even if you have just an ATA. The caveat I have is that sip uri dialing is anonymous -- you do not know who is dialing -- and unless you control anonymous sip, this can lead to trouble - especially in a pbx environment. For ATA users, not as much risk.

FWIW YMMV
potassum6
join:2009-10-11
Montreal, QC

potassum6 to tom thomas

Member

to tom thomas
Thank you for taking time to enlighten me. I have a couple of domains so I;ll give it a whirl.
potassum6

1 edit

potassum6

Member

So I think I did it...I got a domain like clueless.info
I then set up an SRV to john@clueless.info

Question no. 1) Can I just make the john bit up or does it have to be set up as an email address?

I set it to UDP, 0, 0, 5060 and target to the MX record server?
It set up OK.
Question no. 2) now what? I tried registering Xlite to john as username and clueless.info as domain and john@clueless.info...no joy

How do I receive a call then?

Sorry to be thick..
potassum6

2 edits

1 recommendation

potassum6

Member

OK so thanks to W1ve and a sunday of mind numbing nerdyness I have done it. I will attempt to translate from nerdy speak into normal-ish speak from my painful experience so you don't have to...I doubt people here need this as they probably know this well, but I'll post here so I can point others to it if needs be. Basically i didn't realise you need sip service (which is free) to make it work.

Here are idiot-proof steps:

1) if have a web domain like iamthick.biz or whatever, just as you can set up a whole bunch of emails like john@iamthick.biz or 12345@iamthick.biz, etc etc, so you can do the same with SIP URIs. E.g. could have a family all with sip URIs at surname.com. Herman@munster.com, hermanjr@munster.com etc. Same as email.

2) These SIP URIs don't have to be actual emails, but they can be (as well as sip URIs), if you want.

3) To simplify lets assume you get a godaddy .info domain www.godaddy.com/, it will cost $1.11 Lets say the domain munster.info is available

You then need to set up a new SRV record, as instructed here: »www.junctionnetworks.com ··· -records

4) But what's all this onsip business?
Yes the missing info, you need to sign up for a free onsip account. Why? Because that'll make the sip bit work. So set the target as sip.onsip.com and just roll with it.

5) Set up your onsip account, (click on any package and then select to add features), don't select any features and you'll end up with a free account. I'll just say that the features they have are good and resonable but not required for this.

6) Set up the names of the SIP URIs as new users in onsip. So in the example above you'd need herman and hermanjr set up as users herman@munster.info and aso on.

7) then you need to make your onsip account link to your domain you do that by connecting the onsip account to the srv, like it explains here:
»www.junctionnetworks.com ··· p-domain

8)Now if that doesn't work, (and onsip won't accept it if it doesn't), then it could be your webdomain is not live. You need to fiddle about but you can get a page like 'under construction' and then choose free hosting, then go live. It is well hidden probably to make you pay for hosting. So you need to click about for a while until it is live and so you should actually see that underconstruction page when you type in www.herman.info into the address bar.

9) Then you can copy the phone configuration stuff that is listed under the user, say, herman on the onsip account into Xlite soft phone,

it'll look like this:

Address of Record: herman@munster.info
SIP Password: biGH4jGsGFRS86hH
Auth Username: munster_herman
Username: herman
Proxy/Domain: munster.info

Then, assuming the phone registers, you are in business...and can make and receive sip calls to an unlimited number of sip uris of $1.11 a year.

Enjoy
dalrun
join:2008-01-09
Bellingham, WA

1 edit

dalrun to potassum6

Member

to potassum6
SIP URI's are not related to being able to receive calls to your domain, [anything]@yourdomain.tld (top level domain, e.g. .com) will work. You don't need OnSip.com to receive calls to yourdomain.tld. Your phone doesn't need to be registered anywhere to receive calls to your domain.

You don't need a SRV record if 'host [yourdomain.tld]' returns your public IP address (the one your phone is behind). While you can test receiving calls this way by setting up your domain name to point to your public IP address, it's only a good long term solution if you have a fixed IP address.

If you don't have a static IP address, you'll probably want to use a dynamic DNS service so that you have a static hostname (host,domain.tld) for your public IP address. In this case you setup a SRV record for yourdomain.tld that points to your dynamic DNS hostname (which in turn points to your public IP address).

Your domains SRV record points to your phones hostname
_sip._udp.yourdomain.tld. IN SRV 0 0 5060 you.dyndns.com.
which gets translated to your public IP address by your dynamic DNS host's DNS server.
you.dyndns.com. IN A xxx.xxx.xxx.xxx

With your domain pointing to you public IP address you don't need a SRV record because a lookup for sip:[anything]@yourdomain.tld returns your public IP address directly from your domains DNS server.
yourdomain.tld. IN A xxx.xxx.xxx.xxx
You could use a SRV record to specify an alternate port, but I wonder how reliable SRV is (Is there any requirement that SIP clients do a SRV lookup before an A lookup?).
_sip._udp.yourdomain.tld. IN SRV 0 0 5061 yourdomain.tld.

In either case you'll probably need to setup your router to forward port 5060 UDP to your phone/ATA.
potassum6
join:2009-10-11
Montreal, QC

potassum6

Member

Thanks, I barely understand you, but if I follow this is instructions on how to get it to ring at your house? How would you do this on a laptop that you take all over and you have maybe 2 or 3 laptops or on a soft phone on a cell phone?
Would you ever call out on sip uri from a normal phone? I guess it is do-able but is it worth it? I vaguely recall seeing it somewhere.
Do you have this set up at home? What's wrong with the onsip thing? Anyway it all seems pretty flexible at a level that most people don't care to know too much about.
I guess the spirit of the thread is given that in theory you can have lots of free customizable sip URIs for not much money and can call between users for free will it catch on? And the answer would seem to be no unless something happens to make it easier, like web domain sellers added it as a specific feature for a start. Call me buttons are close to it perhaps?
If email progam could detect if an address was email or sip or both and offer the choice to call or email...I don't know, all this has taught me is although I am just about up to slowly figuring it out and I quite enjoy the sad challenge, in the end I'd rather pay voip.ms $3 a month and then bash them about when the service doesn't work...
dalrun
join:2008-01-09
Bellingham, WA

dalrun

Member

Yes, for ringing a single IP address (e.g your home). I didn't mean to diss OnSip. From my side its an unnecessary complication. I use as direct a connection as I can, e.g. GV > G5 (currently better than ipcomms) > my domain and my domain > SIP URI (both via my GV/CC dialer).
potassum6
join:2009-10-11
Montreal, QC

potassum6

Member

No dissing taken, I never used it 'til today and wondered if it was OK, it seemed OK.... So one last question since you seem to have wisdom in this regard....if you point your domain to your house does that mean you don't get a webpage or is this something to do with subdomains or something?
dalrun
join:2008-01-09
Bellingham, WA

dalrun

Member

You could host your domain website at your home as well (most small websites use very little bandwidth).

If you host your website elsewhere and you have a static IP address at home, you could add a subdomain to your domains DNS - an A record that points to your home IP address.
sip.yourdomain.tld. IN A xxx.xxx.xxx.xxx
and a SRV record that points to it
_sip._udp.yourdomain.tld. IN SRV 0 0 5060 sip.yourdomain.tld.
Without the SRV record your SIP URI's would be limited to [anything]@sip.yourdomain.tld ([anything]@yourdomain.tld wouldn't work).

If you don't have a static IP address, you'd need to get a dynamic DNS hostname (free) and create a SRV record that points to it
_sip._udp.yourdomain.tld. IN SRV 0 0 5060 you.dyndns.com.
(same as before, it doesn't matter where your website is hosted).

mgraves1
Premium Member
join:2004-04-05
Houston, TX

2 edits

mgraves1 to tom thomas

Premium Member

to tom thomas
Here's a nice, if little older guide on setting up SIP URI dialing for your domain.

»www.blyon.com/blog/index ··· dialing/

On a semi-related topic. I've been approached recently by companies wanting to place ads on my blog. generally I just tell them no, but as my hosting costs are increasing I'm starting to give them some consideration.

Amongst the conditions I'm putting on even considering their proposal is that they call me to discuss the terms...by way of a sip uri.
potassum6
join:2009-10-11
Montreal, QC

potassum6

Member

Brainwave alert
I saw onsip have a dialler add on for firefox but it doesn't do sip uri

Neither does this nice one
»addons.mozilla.org/en-US ··· /151912/

But I wrote to them and suggested they do it before someone else does. Now that would be great, you see a someone@something.com hyperlink, you hover mouse over & it says call or mail, or better it knows if both are available, click and call.
Then sip URI might take off, no numbers or letters. I guess greasemonkey could make it work....anyone can do that here?

mgraves1
Premium Member
join:2004-04-05
Houston, TX

mgraves1

Premium Member

Click for full size
M.OnSIP Portal Showing Dialing By SIP URI
said by potassum6:

Brainwave alert
I saw onsip have a dialler add on for firefox but it doesn't do sip uri
While the OnSIP FF plug-in may not handle SIP URI, their My.OnSIP portal definitely does. The function that invokes a call will take a SIP URI and then spawn a pair of outbound call legs, bridging the two when one answers.

I use this all the time as it's a great way to dial ad hoc SIP URI without programming it into my Polycom phone.
w1ve
Premium Member
join:2007-12-28
Hancock, NH

w1ve

Premium Member

I was away from this topic for some time... Just want to make a few comments:

- SIP URI dialing CERTAINLY does not require a SIP account with Anyone!

- SIP URI dialing is peer-to-peer --geek speak for "My computer dialing your computer directly".
- All the DNS and SRV record stuff is geek magic so that you do not have to actually know the IP addresses of the computer that you trying to connect to.

Let's say you have a voip.ms DID. That DID has the SIP URI
NXXXXXXXXX@servercity.voip.ms.

If you have a softphone that can dial SIP URIs directly, you can enter that above URI and you'll be connected. It's a direct connection between you and the voip.ms server.

Everyone gets confused about SIP URI dialing, because most times you register your ATA or softphone with a service provider. As soon as you register, you are telling your phone that all calls get routed via that provider, be it your Asterisk Box, OnSIP, PBXes, or anyone else. Once you are registered,
than you are using the provider dial plan. If the dial plan does not support SIP URI dialing, it's not going to work.

Counterpath has broken SIP URI dialing in the free version of xLite, because it won't allow you to dial unless it is registered with a server! Zoiper seems to work. It seems more and more softphones won't allow SIP URI dialing.

tom thomas
@qwest.net

tom thomas

Anon

said by w1ve:

I was away from this topic for some time... Just want to make a few comments:

- SIP URI dialing CERTAINLY does not require a SIP account with Anyone!

- SIP URI dialing is peer-to-peer --geek speak for "My computer dialing your computer directly".
- All the DNS and SRV record stuff is geek magic so that you do not have to actually know the IP addresses of the computer that you trying to connect to.

Let's say you have a voip.ms DID. That DID has the SIP URI
NXXXXXXXXX@servercity.voip.ms.

If you have a softphone that can dial SIP URIs directly, you can enter that above URI and you'll be connected. It's a direct connection between you and the voip.ms server.

Everyone gets confused about SIP URI dialing, because most times you register your ATA or softphone with a service provider. As soon as you register, you are telling your phone that all calls get routed via that provider, be it your Asterisk Box, OnSIP, PBXes, or anyone else. Once you are registered,
than you are using the provider dial plan. If the dial plan does not support SIP URI dialing, it's not going to work.

Counterpath has broken SIP URI dialing in the free version of xLite, because it won't allow you to dial unless it is registered with a server! Zoiper seems to work. It seems more and more softphones won't allow SIP URI dialing.
all of this is correct. i would like to add that the whole concept of SIP URI calling in general and direct P2P SIP in particular has gone straight downhill through the history of SIP starting with the early days of services like pulver's FWD.

we have moved from a radical breakaway from the traditional telecom to essentially taking on a similar model to those companies.

at this point in my opinion the greatest thing that could happen would be for another standard such as jabber/jingle used in gtalk to start to take off in a bigger way giving society an open source option that does not get taken over with PSTN interconnection and stands on its own as a P2P standard that allows voice communication having nothing to do with traditional phone companies. if deployed on the popular mobile phone platforms such as android and perhaps even the iphone this could gain some real traction that gives an open alternative to skype.

mgraves1
Premium Member
join:2004-04-05
Houston, TX

mgraves1

Premium Member

said by tom thomas :

all of this is correct. i would like to add that the whole concept of SIP URI calling in general and direct P2P SIP in particular has gone straight downhill through the history of SIP starting with the early days of services like pulver's FWD.

we have moved from a radical breakaway from the traditional telecom to essentially taking on a similar model to those companies.
It's true that the business model and the technological model are at odds. We have largely maintained the business model of what some like Aswath Rao (»www.mocaedu.com/mt/) have called "trapezoidal voip" despite that fact that service providers are largely unnecessary.

However, the reality of SIP and NAT mean that a "registrar" is often a good thing, even if not absolutely required.

Using OnSIP merely as my SIP Registrar means that I don't need to accept incoming SIP sessions from just anything at all. All SIP signaling for incoming call setup is initiated by an OnSIP server to my phones.

It makes it easier to secure my network as I don't have any ports forwarded, etc. All traffic is through pinholes transiently established with known source.

It's not perfect, but it's better that complex port management at my network edge, or requiring me to have an SBC for just a few phones behind a router.

There are underlying issues of identity assertion and proof that remain to be comprehensively addressed if we are to have true peer-to-peer voip.

Related links:
Aswath Rao on VUC
»www.voipusersconference. ··· ath-rao/

Supporting Slides
»www.slideshare.net/aswat ··· -is-evil

tom thomas
@qwest.net

tom thomas

Anon

i am not against the server model if the business/technological model were to have developed more like email. but rather the analogy is more like this:

you type an email and the provider stuffs it into an envelope and than charges you for the necessary postage. at the other end the paper gets scanned and delivered electronically to the receivers inbox. lets forget for a moment about the time delay that would be involved and we see the business/technological model that has been taken on by VOIP operators providing 'land line replacement' services.

kieranmullen
Premium Member
join:2005-12-12
Portland, OR

kieranmullen to w1ve

Premium Member

to w1ve
Old Multi Sip Version Client

»360oregon.com/files/X_li ··· 103a.exe
»360oregon.com/files/X_li ··· 7_28.exe
»360oregon.com/files/X_li ··· 1_11.dmg
said by w1ve:

Counterpath has broken SIP URI dialing in the free version of xLite, because it won't allow you to dial unless it is registered with a server! Zoiper seems to work. It seems more and more softphones won't allow SIP URI dialing.

Anpox
@llamaz.com

Anpox to tom thomas

Anon

to tom thomas
I tell my relatives in Chicago to get me at: 1-630-405-0730 ext. *0111xxx

where xxx is my sipbroker alias. There was some complaining, but they are used to it by now and have no problems.

Maybe some universal gateway number, followed by an extension, is a way to ease landline users into the voip world.
GraysonPeddi
Grayson Peddie
join:2010-06-28
Tallahassee, FL
Ubiquiti EdgeRouter PoE
Ubiquiti UniFi AP-AC

2 edits

GraysonPeddi

Member

Hi everyone. It seems I got inbound SIP URI calling to work, but due to the way how FreeSWITCH works, I had to append port 5080 to the end of my SIP address like this:

sip:yourname@yourdomain.com:5080

If I don't append 5080, it's not going to work because the internal profile has a UDP port of 5060 and the external profile has a UDP port of 5080.

I have my phone number set up in e164.org and tell it to use my SIP address. Once I receive a phone call, I answered it, and although I hear silence, it works.

tom thomas
@qwest.net

tom thomas to Anpox

Anon

to Anpox
said by Anpox :

I tell my relatives in Chicago to get me at: 1-630-405-0730 ext. *0111xxx

where xxx is my sipbroker alias. There was some complaining, but they are used to it by now and have no problems.

Maybe some universal gateway number, followed by an extension, is a way to ease landline users into the voip world.
while these gateways may help ease in some they also serve to slow down the adoption of going to a direct P2P world that the big telecom have no part in.

Aswath Rao
@verizon.net

Aswath Rao to mgraves1

Anon

to mgraves1
Let me clarify my point against Trapezoidal VoIP. I am questioning the need for the Outbound Proxy. Most of us don't use a proxy while using HTTP; why SIP needs to. Agreed that there may be policy requirement that would require Outbound Proxy; but not as a default.

I do not advocate serverless or P2P systems. Let us have registrars and inbound proxies. That is all fine.

Such an architecture will require some things that generally deployed/used SIP don't have. It absolutely requires URI based addressing because the originating client will not know which Proxy to contact with just a telephone number. Second, the originating caller must be able to authenticate to a third party Proxy. This means we need OpenID like authentication mechanism.
w1ve
Premium Member
join:2007-12-28
Hancock, NH

w1ve to kieranmullen

Premium Member

to kieranmullen
said by kieranmullen:

Old Multi Sip Version Client

»360oregon.com/files/X_li ··· 103a.exe
»360oregon.com/files/X_li ··· 7_28.exe
»360oregon.com/files/X_li ··· 1_11.dmg
said by w1ve:

Counterpath has broken SIP URI dialing in the free version of xLite, because it won't allow you to dial unless it is registered with a server! Zoiper seems to work. It seems more and more softphones won't allow SIP URI dialing.
Thanks!
OmagicQ
Posting in a thread near you
join:2003-10-23
Bakersfield, CA

OmagicQ

Member

I found the following on this page »www.fredshack.com/docs/sip.html
quote:
X-Lite can also be used to make direct IP calls without using an SIP proxy to register. To do this, here's how to register the local host in the SIP Account Settings dialog:

Account tab: Display name = Whatever you want, User name = whatever you want, Password = none, Authorization user name = none, Domain = local IP of this computer, Domain Proxy : Uncheck "Register with domain" and select target domain

Topology : Port used on local computer = 5060-5061. Make sure no other application is listening on those ports, as X-Lite won't say anything if it cannot bind to the ports.

Next, open the Contacts Drawer (little arrow on the right-side of the main window), and add an account in the Contacts section of Type = Softphone and Address = remotename@remoteIP, eg. john@192.168.0.1 if the remote X-Lite user registered as john and is running the app on host 192.168.0.1. Double-click on this account to make a call.

If you get a "488 Not Acceptable Here", make sure X-Lite/EyeBeam has at one codec in common with the remote party (eg. if the PBX uses G711 uLaw and X-Lite only uses G711aLaw, it will fail working. More information in EyeBeam Troubleshooting Guide.