[SIPForum-discussion] UAs
Manpreet Singh
msingh at ibasis.net
Sat Mar 4 18:15:42 UTC 2006
Richard
Thanks for the response. Yes I know about the reversed field, I think I
copied the example from 2916 instead. So I guess SRV is the right way to go
about it.
Just on the side note, is there any document available explaining the reason
as to why the fields were reveresed? I have tested severel Uas, including
cisco UA ( GWs and IPIP GWs) and they fail the call with new implementation,
which is E2U+SIP. If you can share any document explaining why this change
was done, I would really appreciate it
Thanks
Manpreet
-----Original Message-----
From: Richard Shockey [mailto:Rich.Shockey at neustar.biz]
Sent: Saturday, March 04, 2006 1:09 PM
To: Manpreet Singh
Cc: Maddox, Sean (MVNSO Solutions Mgr); Henning Schulzrinne; Klaus Darilion;
discussion at sipforum.org
Subject: Re: [SIPForum-discussion] UAs
Manpreet Singh wrote:
> 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!" .
IN 3761 BTW the E2U and enumservice field is reversed
as in E2U+sip
> 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 lost of
> servers and then send the INVITE to those servers. My assumption is
> that the client would so SRV first because doing a A record lookup
> would not result in the actual termination proxy or endpoint.
>
> Correct me if I am wrong.
That is correct.
>
> 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* <_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
>
> 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
>
> ----------------------------------------------------------------------
> --
> *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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>
>
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>
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--
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Director - Member of Technical Staff NeuStar Inc.
46000 Center Oak Plaza - Sterling, VA 20166
sip:rshockey(at)iptel.org sip:57141(at)fwd.pulver.com
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