[SIPForum-discussion] UAs

Manpreet Singh msingh at ibasis.net
Tue Feb 28 04:39:14 UTC 2006


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

_______________________________________________
discussion mailing list
discussion at sipforum.org
http://sipforum.org/mailman/listinfo/discussion
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20060227/0f3d0d9a/attachment-0001.html>


More information about the discussion mailing list