[SIPForum-discussion] UAs

Manpreet Singh msingh at ibasis.net
Thu Mar 2 16:42:46 UTC 2006 

Hi
 
Anything on this? Any input is appreciated. Different UAs have implemented
this differently ( some do SRV and some do A record for the domain) so not
sure what should be the right bahviour. I feel SRV shoul be done but then
its just my opinion.
 
 
Thanks
Manpreet

  _____  

From: Manpreet Singh 
Sent: Tuesday, February 28, 2006 6:52 AM
To: 'Maddox, Sean (MVNSO Solutions Mgr)'; Henning Schulzrinne; Klaus
Darilion
Cc: discussion at sipforum.org
Subject: RE: [SIPForum-discussion] UAs


Sean
 
True but these dont happen when using NAPTR for ENUM services. In the
example you gave, the "s" flag tells the client that the next lookup is a
SRV lookup. For ENUM response, most of the time in the replacement string
would carry a sip URI and the flag is "u". ( terminal ) Check the  example
below:
 
$ORIGIN 2.1.2.1.5.5.5.0.7.7.1.e164.arpa.
IN NAPTR 100 10 "u" "sip+E2U"  "!^.*$!sip:information at tele2.se!"     .
IN NAPTR 102 10 "u" "mailto+E2U" "!^.*$!mailto:information at tele2.se!"  .

Now in the above example, would the UA be expected to do a SIP INVITE to a A
record of tele2.se or would be it do a SRV lookup for _sip._udp.tele2.se. (
assuming it can only do UDP), receive a list of servers and then send the
INVITE to those servers. My assumption is that the client would  do SRV
first because doing a A record lookup would not result in the actual
termination proxy or endpoint.
 
Correct me if I am wrong.
 
Thanks
Manpreet
 

  _____  

From: Maddox, Sean (MVNSO Solutions Mgr) [mailto:sean.maddox at hp.com] 
Sent: Tuesday, February 28, 2006 1:33 AM
To: Manpreet Singh; Henning Schulzrinne; Klaus Darilion
Cc: discussion at sipforum.org
Subject: RE: [SIPForum-discussion] UAs



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 < <mailto:sean.maddox at hp.com> 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 

As always, please let me know if you prefer not to receive these emails from
me. For more information regarding Hewlett-Packard's policy or to obtain
contact information, please see our privacy statement at:
http://thenew.hp.com/country/us/eng/privacy_intent.html
<http://thenew.hp.com/country/us/eng/privacy_intent.html> 


  _____  

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
<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://
<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 <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:// <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
<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
<http://sipforum.org/mailman/listinfo/discussion>  

_______________________________________________ 
discussion mailing list 
discussion at sipforum.org 
http://sipforum.org/mailman/listinfo/discussion
<http://sipforum.org/mailman/listinfo/discussion>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20060302/9c71a867/attachment-0001.html>

More information about the discussion mailing list