[SIPForum-discussion] UAs

Tue Feb 28 06:33:18 UTC 2006

Manpreet,

My understanding is that the NAPTR response isn't a SIP URI but instead
should contain at least 3 records each of which identifies a service
(which itself defines both service & protocol) and an associated target
for that service.  Service in the context of SIP NAPTR records means
either SIP (non-secure) or SIPS (secure) while protocol in this context
means TCP, UDP or SCTP.  The combination of protocol and service being
represented by the DNS response as SIP+D2U (SIP over UDP), SIP+D2T (SIP
over TCP), SIPS+D2T (secure SIP over TLS over TCP) and SIP+D2S (SIP over
SCTP).  The SIP client processes these NAPTR records with an order of
preference for selection of SIPS+D2T (secure & reliable transport),
SIP+D2T (un-secure & reliable) and finally SIP+D2U (un-secure &
un-reliable).  The replacement value associated with each NAPTR record
identifies the value to issue the SRV DNS request against.

>From RFC 3263, the NAPTR response might contain the following values:

   ;          order pref flags service      regexp  replacement
      IN NAPTR 50   50  "s"  "SIPS+D2T"     ""  _sips._tcp.example.com.
      IN NAPTR 90   50  "s"  "SIP+D2T"      ""  _sip._tcp.example.com
      IN NAPTR 100  50  "s"  "SIP+D2U"      ""  _sip._udp.example.com.

The client is supposed to first choose a service and then issues a DNS
SRV request using the replacement value (for example
_sips._tcp.example.com) which in turn would yield a target of something
to the effect of sips-service.example.com and a port.  If a numeric IP
address is returned instead then the client uses the address, if no port
is returned then the default is assumed.  If a non-numeric value is
returned the client should issue a DNS A or AAAA request against the
target value to resolve it.

That is at least the way I read, and re-read, things.

Thx - Sean 

Sean P. Maddox <sean.maddox at hp.com <mailto:sean.maddox at hp.com> >
IP Communications Solutions Manager - HP Americas 
Mobility, Voice & Network Solutions
The Hewlett-Packard Company 
+1 817.898.0218 
sip:sean.maddox at hp.com 

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________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Manpreet Singh
Sent: Monday, February 27, 2006 10:39 PM
To: Henning Schulzrinne; Klaus Darilion
Cc: discussion at sipforum.org
Subject: RE: [SIPForum-discussion] UAs
Importance: High

Hi 

I have a question regarding the NAPTR implementation. For a NAPTR
response which is a SIP URI, is the UA meant to do a simple A record
lookup and initiate a request or should be really doing a SRV lookup and
then follow the A record lookup. I have seen different UAs behaving
differently ( some do SRV first and some do A record lookup ) so was
curious as to what would be right behaviour/implementation. 

If not in spec then is it purely on the discretion of a UA
implementation? I would usually expect a host portion to be a domain in
the response ( eg bob at abc.com ) which would host proxies, hence SRV
would make sense. Opinions/suggestions??

Thanks 
Manpreet 

-----Original Message----- 
From: Henning Schulzrinne [mailto:hgs at cs.columbia.edu] 
Sent: Monday, February 27, 2006 9:37 PM 
To: Klaus Darilion 
Cc: discussion at sipforum.org 
Subject: [SIPForum-discussion] UAs 

The list is quite helpful. I'm trying to flesh out the list at http://
www.cs.columbia.edu/sip/ua.html with additional data on modern (3261),
actively-maintained client, i.e., you would want to recommend to a
friend new to SIP. It would be helpful if those who have used (or
written) clients can provide information about their UA, such as

- SIP features: TCP, TLS, NAPTR 
- audio codecs, including RFC 2833 support 
- video codecs 
- presence - basic? rich? XCAP? 
- license 
- any additional remarks such as restrictions or particular features 

I'm focusing on UAs that can be configured for any proxy, not just those
that are part of a service. 

Thanks. 

On Feb 27, 2006, at 6:04 AM, Klaus Darilion wrote: 

> Bill Nash wrote: 
>> <mailto:discussion at sipforum.org> Hi! 
>> I am beginner for softphone applications. I want to implement a 
>> simple softphone between two PC (end-to-end) with SIP using C++ 
>> language on Linux. I need some advice about tutorial, API and 
>> whatever you want to say about  it. 
> 
> 
> Hi Bill! 
> 
> First, I would not write a new SIP phone, but extend existing SIP 
> phones. You can find a list of free sip phones at: http:// 
> www.pernau.at/kd/voip/bookmarks-sip-phones.html 
> 
> If you really want to write a new SIP phone, check out the existing 
> SIP stacks: 
> http://www.pernau.at/kd/voip/bookmarks-sip-stacks.html 
> 
> regards 
> klaus 
> _______________________________________________ 
> discussion mailing list 
> discussion at sipforum.org 
> http://sipforum.org/mailman/listinfo/discussion 

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