[OpenSIPS-Users] Basic doubt of sip routing

Mon Feb 14 09:09:45 CET 2011

Toyima,
The client does NOT have to use NAPTR or SRV. It can use regular A or
CNAM records as well or even just an IP. (unless it is a half baked
client)
SRV and NAPTR are special DNS records that can be used tell the client
what services are available and where and how to connect to them. If
your starting out, don't worry about them. When you start to get into
failover and load balancing, then give them a look over.
If other service is providing them for opensips to use, nothing
special to do for opensips, it does that all automatically.

The most common use of opensips it to take a call request to a phone
number and decide where the next place to send that call request. A
super simplified two steps:
1 lookup if the called number belongs to one of your customers. If so
send it to them.
2 If no above, then lookup if the source of the call is one of your
customers. If so send the call through a carrier that can get it to
the owner of the number. (basically PSTN -- regular telephone
networks)

There are many ways that opensips can accomplish those tasks. As
Adrian said about choosing where to send it can be done, there are
multiple ways to check where the call came from and if want to send
the call on, like authentication via DB or radius, static source IP in
script, a module like dynamic_routing, or even a custom sql query.
It all depends on your needs.
If your just getting started, there are some good recommendations in
some fairly recent email threads for where to start using what
resources.

Dave

On Fri, Feb 11, 2011 at 5:16 AM, Toyima Dias <toyimads at gmail.com> wrote:
> COOL Adrian...many thanks for your kindly answers...by the way, i've checked
> on the rfc that the client must use NAPTR and SRV to resolve domains!
>
> 2011/2/11 Adrian Georgescu <ag at ag-projects.com>
>>
>> If your SIP device support dialing only phone numbers, you need a
>> translation mechanism, this you can implement in the SIP proxy. You can use
>> standard ENUM (http://www.faqs.org/rfc/rfc3764.txt),  local database
>> lookups, configuration logic to translate the number into a fully qualified
>> SIP address or many other methods, there are plenty of OpenSIPS modules that
>> do such phone number translation.
>> Adrian
>> On Feb 11, 2011, at 2:04 PM, Toyima Dias wrote:
>>
>> create...got it...that means that if i have a phone registered in proxy A,
>> and i want to call userB, A has no idea where B resides, at all...how does A
>> know the domain of B? he must put in the RURI of the invite
>> userB at domain_of_b, right? how does A knows the domain of B? does A must
>> press in the phone: usearB(a number)@DOMAIN_OF_B?
>>
>> sorry for many questions, keep reading the rfc...
>>
>> 2011/2/11 Adrian Georgescu <ag at ag-projects.com>
>>>
>>> SIP routing works exactly like email. How did you know to email this
>>> list?
>>> Adrian
>>>
>>> On Feb 11, 2011, at 1:42 PM, Toyima Dias wrote:
>>>
>>> Thanks Adrian...
>>> So...how does ALice now that bob is in the biloxi.com domain? per the rfc
>>> 3263 section 4 (client usage) the ua must use DNS to determine where to send
>>> a call...but i have a softphone righ now, and i'm trying to make a call like
>>> this:
>>>
>>> 234560 at proxy2.com (inserted by me), not just puting the number....maybe
>>> i'm saying something crazy, but i'm confuse!
>>>
>>> 2011/2/11 Adrian Georgescu <ag at ag-projects.com>
>>>>
>>>> The SIP proxy lookups up the domain part, what appears after the @ sign
>>>> before any parameters separated by ; if is an IP address like in your
>>>> example you do not perform a DNS lookup you just send the packet there.
>>>> In the request URI you must put the address of the remote end, not your
>>>> own address. In your example user A calls user 264512380973 on the same
>>>> Proxy and not a remote one.
>>>> See the illustrated examples from:
>>>> http://www.tech-invite.com/Ti-sip-ex3261.html
>>>> Adrian
>>>> On Feb 11, 2011, at 1:17 PM, Toyima Dias wrote:
>>>>
>>>> Adrian, i'm checking the rfc...but even i have a question...when UA
>>>> sends an INVITE to it's proxy to a phone for example (obviously not
>>>> registered on the proxy), the proxy will check the RURI of this invite and
>>>> it will se the following:
>>>>
>>>> user A sends the invite to its proxy : INVITE
>>>> sip:264512380973 at 172.30.140.57;transport=UDP SIP/2.0 (172.30.140.57 is the
>>>> IP of proxy A)
>>>> Where does the DNS takes part? the domain is it's ip address...i'm quite
>>>> confuse, any help would b e appreciated
>>>>
>>>> Thanks
>>>>
>>>> 2011/2/11 Toyima Dias <toyimads at gmail.com>
>>>>>
>>>>> Thanks Adrian...reading the RFC3263!
>>>>>
>>>>> Thanks!
>>>>>
>>>>> 2011/2/11 Adrian Georgescu <ag at ag-projects.com>
>>>>>>
>>>>>> The proxy is using DNS to lookup the destination server.
>>>>>> Google for RFC 3263
>>>>>> Adrian
>>>>>>
>>>>>> On Feb 11, 2011, at 10:19 AM, Toyima Dias wrote:
>>>>>>
>>>>>> Hello community,
>>>>>>
>>>>>> I have a doubt, how does a SIP Proxy (OpenSIPS) would handle a call
>>>>>> for a domain that he doesn't now? i mean...user A is registered in proxy AA,
>>>>>> if A wants to call to another user in another domain (not registered in the
>>>>>> Proxy AA) how does this proxy should handle the call? how does he now where
>>>>>> to send this call?
>>>>>>
>>>>>> Any clarification would be appreciated :)
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>>>>>>
>>>>>>
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>>>>>
>>>>
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