Can we make SIP and H323 Annex O calls from free Jabber Video beta?

Unanswered Question
Mar 12th, 2012

With the free Jabber Video client, I see that I can not dial by IP Address or by <H323_ID/Alias>@<IP_Address>.  Does this service only allow for SIP calls and NOT for calls made to H.323 endpoints (using Annex O dialing)?

I have this problem too.
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crunkisized Mon, 03/12/2012 - 14:30

Hi Rick - thank you for the reply.  Is there any limitation/requirement regarding SRV or A record lookups on the "domain.com" piece of the puzzle?  Meaning, will the VCSs on the backend only do a specific SRV query for "_sipcs", or should I expect that it tries that first, and then should try "_h323cs._tcp.domain.com".

Thanks again for the feedback.  This will help me better articulate with customers that we need to communicate with.

Brett

Hoan Mai Mon, 03/12/2012 - 16:41

Brett,

CJV will try sip 1st then h323 as the order of preference.  The SRV record needs to be inplace in order the call to

reach the domain properly.


Rick Mai

crunkisized Mon, 03/12/2012 - 19:41

Hi Rick / Jens -

Again, thanks for the replies.  At this point, guess I would just ask if we could verify whether SIP to H323 interworking is turned ON or not ... I have not been able to connect to H323 at this point, and Jens sounds like he may have tried, but got no media.  We have the proper SRV records setup for "_h323cs._tcp.our_domain.com" along with the subsequent A records for the reply.

Thanks,

Brett

Jens Didriksen Mon, 03/12/2012 - 20:51

Yup, unless Cisco has made some changes very recently, which I will find out when I test again from home in an hour so...

My finding has been calling h323 endpoints, using h323 addresses, the call will connect, but I get no media at all.

Connecting to same end-point using its SIP address works like a charm.

Connecting from the same location using the not-free version, 4.3 - both the h.323 address and the SIP address works fine, no problems with media.

Edit: Turning off SIP on the end-point makes no difference, call drops after a few seconds.

/jens

Jens Didriksen Mon, 03/12/2012 - 23:31

Just tried from home with "not free" version 4.3 on Mac registered to my VCS-E

free version 4.4 on PC registered to Cisco cloud.

4.3: Both SIP and h323 addresses connects fine and media is also fine.

4.4: SIP address connects fine, media is fine.

H323 address is treated as a SIP--SIP instead of SIP---H323 by 4.4 and fails. Turning off SIP on the endpoint makes no difference.

(Guess I can overcome that by changing or adding a search rule and/or transform, but that's really neither here nor there as this is not important for us.)

Did some more testing the other day, yes, I'm a sucker for punishment - anyways, turned off SIP on the VCS-E, and calls from "free" 4.4 are now denied, which makes sense since it's not trying h.323 instead.

/jens

Jens Didriksen Mon, 03/12/2012 - 15:25

Edit: I am beginning to think h323 Annex O from free Jabber is not working for me as I have SIP enabled on the VCS' - might try turning that off and see if this free version will then go h.323 instead - turning off SIP on the endpoint makes no difference at all. Not that it matters as all of our endpoints do both h323 and SIP.

This free version does indeed support alias@domain as well as alias@IP_address.

We have SRV records in place for both H323 _ls and _cs - and SIP SRVs for udp, tcp and tls - works well.

We use the "not free" version internally, and with this I can call just about anything I want; whether it is SIP, H323, H320 - not because the client is capable of this - which it isn't, but because the VCS can.

Edit: What I have just discovered is that this "free version" will not play properly with the "not free" version; the call will connect, but no video and only occasionally one way audio -this is not good Cisco

Edit: Cisco is indeed aware of the above, it appears to work ok though for users in USA/Canada.

Note: For Alias@IP_address to work, a pre-search transform must be in place on the VCS-E which transforms Alias@IP_address to Alias@domain. Then it depends upon the search rules you have in place.

/jens

crunkisized Wed, 03/21/2012 - 09:21

Hi Jens / Rick -

Again, really appreciate the feedback - but I want to take the "free" "not free" client out of this - I've read through a couple times and I'm a little confused.  I'll simplify the scenario:

If I'm at home, or Starbucks (taking enterprise corp FWs out of the picture) and I launch the Free Cisco Jabber for Video client - it registers to the cloud and I'm ready to make a call.  I try to H323 AnnexO Dial a system like so "@mydomain.com" where I have the proper "_h323cs._tcp.mydomain.com" SRV record in place, as well as subsequent A records in place (to resolve the svr hostname in the SRV response).  No SIP in this case.  The call is not connecting ...

Is this an issue with the Free Jabber service, or is working as designed?

Edit: I just noticed that when I add a favorite, it prepends "sip:" to the front of the URI after I save it and then view it.  I manually changed this to "h323:" and it appears to have saved it, but it still does not work.

Thanks,

Brett

Message was edited by: Brett Wiggins

Jens Didriksen Wed, 03/21/2012 - 16:02

The more I look at this the more confused I get as well, see my last post in this thread:

https://supportforums.cisco.com/thread/2136926?tstart=0

It is, generally speaking, normal behavour that it prepends "sip" to the front of the uir, but whether or not this normal behavour for this particular version is another story all together.

If it in fact looks for h.323 SRV records, then you'll need a SRV record in place for _h323ls_udp for port 1719.

/jens

crunkisized Fri, 03/30/2012 - 09:01

HI Rick / Jens -

I am still questioning whether this free Jabber video client 'service' is able to dial H323 endpoints.  I have still not been able to connect to H323 endpoints - SRV for _h323cs_ is setup ... I do not believe I would also need _h323ls_ as well.

Also, when I save a Favorite, I see that it prepends the URI with "sip:".  I have tried (for the heck of it) to change this to "h323:", and it will take this and save it, but does not appear to send out.

I've also tried @ as well as @, still no success.

Can Cisco confirm that calling H323 endpoints is, in fact, supported?

Thanks,

Brett

Jens Didriksen Fri, 03/30/2012 - 15:17

_ls and _cs are, as you know, specific for H.323, whether you need both in this particular case, I do not know, however, if you want h,323 end-points being able to call you using alias@domain etc, then you need to have both in place to allow both registered and unregistered end-points to connect, as _cs is used by end-points which are not registered to a GK, whereas _ls is used by GKs on behalf of end-points registered to it.

I have also not been able to connect with "free Jabber" to "h.323 only" end-points, i.e. when connecting to a public end-point which is not registered to GK nor a SIP server, and which has only h.323 enabled, by dialling it's IP address in the format jabbervideo@IP_Address, the call will not connect.

However, once I enable SIP on the same end-point, the call connects fine.

Did some more testing whith a couple of MXPs at work behind our VCS-E, again, with SIP off, calls fail, SIP on, calls connect.

Doing the same with our "enterprise" version works fine regardless of SIP being on or off on the end-point thanks to h.323/sip interworking in place on our VCS-C.

All of the above leads me to conclude calling H.323 only end-points are not supported by the "free" version - at least not for now - and at least with our particular deployment

But then things have a tendency to change pretty quick around here...what's not working one day is working the next...

/jens

Lucas Phelps Sun, 04/01/2012 - 23:49

I also have tried numerous H.323 endpoints without success -- only with SIP endpoints.

I believe it is a limitatation Cisco has on their free cloud-VCS that the free client registers to.

mdehaven Fri, 04/06/2012 - 12:27

Jabber is a SIP client only and cannont talk to h323 endpoints without something like a VCS to interwork the media.  Interworking needs to be enabled local to the dialing domain of the h323 endpoint you are calling.  Jabber cloud service does not provide interworking and that is probably a good thing as it would require the media to be hairpined at the Jabber.com sip proxy.

npsmonterey Tue, 04/24/2012 - 13:30

It was the paragraph below, from the release notes, that had me confused. The first sentence tells me I should be able to call our 323 endpoints behind a Tandberg BC. The second sentence seems to qualify that but only refers to Cisco endpoints, and your reply here says definitively that free Jabber can only make SIP calls.

In that case, what should we expect when dialing endpoints behind other OEM's interworking products (Polycom, etc)?

"Interoperability: Place calls from  Jabber Video to any  standards-based video endpoint, provided it is  DNS-routable. You can  call Cisco TelePresence endpoints in a business  network (with VCS  Expressway configured), Callway, or other Jabber  Video users, among  others."

https://supportforums.cisco.com/docs/DOC-20940

Martin Koch Sat, 05/05/2012 - 04:54

Hi Matthew!

Not sure if your month old posting is already outdated, but I can make successfull calls

from a cisco jabber video client to a h323 only endabled domain which pop up from the cisco

vcs to my vcs on port 1720 as a pure h323 call incl. proxied media.

Which would also reflect what Rick said in the beginning of this thread.

Besides that I would assume that "Calls to unknown IP addresses" is simply set to off

Martin

Jens Didriksen Sat, 05/05/2012 - 06:07

Other people are now reporting the same over in the VTCTalk forum, so looks like Rick was on the money all along.

Took me while to realize it was my SIP SRV records which was preventing me from calling any of my end-points using H.323, but the penny finally dropped.

/jens

Jens Didriksen Fri, 04/06/2012 - 15:48

Yup, I think we all know it's a SIP client - however, what confused the issue a bit was an earlier statement by a Cisco rep saying "CJV will try sip 1st then h323 as the order of preference" - which could suggest that interworking was indeed in place.

/jens

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Posted March 12, 2012 at 12:10 PM
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