Direct SIP calls, without registration to CUCM, from Unified IP Phones

[tenaro.gusatu.novici]

Is it possible to use Cisco IP Phones, for example 6900 or 8900 or 9900 series, to send INVITE directly to manually predefined IP address (let's say SIP proxy/registrar) without being registered to CUCM? In other words, I would like to use Cisco IP phone as standard SIP endpoint. In addition, it will be nice to know are there any other options I can manually configure, for example to specify if outgoing call will use early offer or delayed offer.

[Ayodeji Okanlawon]

I do not believe you can do that. Cisco phones need a call control agent before they can function. Hence you will need some form of call agent to regster them with

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[Leo Laohoo]

Is it possible to use Cisco IP Phones, for example 6900 or 8900 or 9900 series, to send INVITE directly to manually predefined IP address (let's say SIP proxy/registrar) without being registered to CUCM? In other words, I would like to use Cisco IP phone as standard SIP endpoint. In addition, it will be nice to know are there any other options I can manually configure, for example to specify if outgoing call will use early offer or delayed offer.

Yes.  It's possible.   But it ain't easy.  Cisco has, since the advent of 7970 and 79X5, made sure this function is NOT DOCUMENTED for commercial reason.

You need to make sure you enter your Voice Service Provider details in the SEPmacaddress.xml.cnf correctly.  Pay close attention to the following details from your VSP:

1.  Enable or disable NAT;

2.  Incoming and Outgoing proxy;

3.  Codec supported;

4.  NTP/SNTP details;

5.  Username/Password provided to you.

In some cases, the VSP will only accept ONE call registration request.  For instance, you want to have two phones trying to register to a single SIP account, this is significant challenge getting incoming calls as the calls will ring in a random or round-robin fashion.

If you want to have multiple phone units then you need to consider getting a SIP server.  Daunting as it may seem, it's not.  I got my FreePBX/Asterisk in a Raspberry Pi (you can use a Beagle Bone as an alternative to a Raspberry Pi) working in 2 days (while I was sick with a bad case of flu).

NOTE:  Another thing, very important, query your VSP if they require ALG to be enabled/disabled.  Don't confuse ALG with NAT.  They're very different.

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[tenaro.gusatu.novici]

Thanks Leo,

any chance you can write detailed instructions (what must be configured on phones) to make simple direct call between two Cisco IP phones (like two 9900)? So, no VSP, I simply want to understand what needs to be configured to make direct call.

Cheers,

Tenaro

[paolo bevilacqua]

Cisco phones do not support direct calls.

[tenaro.gusatu.novici]

Thanks Paolo!

Any chance you know where I can find that statement on CCO? Just so it is official. I have an old Tandberg client and they reached same conclusion (SIP on Cisco phones is "proprietary" SIP as they call it). So I was simply looking for a second opinion. Seems like SIP in Cisco world is so broken that even the most basic defiinition of a SIP as peer to peer protocol can't be applied to Cisco phones.

[Leo Laohoo]

Any chance you know where I can find that statement on CCO?

Google it.  It's very well known to voice integrators.

any chance you can write detailed instructions (what must be configured on phones) to make simple direct call between two Cisco IP phones (like two 9900)? So, no VSP, I simply want to understand what needs to be configured to make direct call.

Ok, let me put it in a different method:

Two phones, in the same IP subnet.  How do they know what codec?  How do they know what is their extension numbers?  How do they know what ports to use?  UDP?  TCP, etc.

Cisco phones are "dumb" units.  They need SOMETHING to manage their configuration and push their configurations.    This something is a call server.  This call server can be housed in your location and/or your VSP.

If a call server is housed in your location, you also need to ensure that YOUR PHONES get to use your facilities and not some stranger plugging their phone to any of your wall port and making 1-900 or internation calls on your money.

If you want to run an extension number so both phones can call each other internally, then you still need this call server.  You can't just go to e-bay and say, "I'll take two of `em, thanks!" and then, presto!  your phone works.  No they won't.  Not with a Cisco enterprise-grade phone.

In my first post (above), I've provided you with cheap alternatives to a full-blown Cisco CUCM/CUE.  All in all, my Asterisk/FreePBX setup at home cost me

Have a look at this link below:

Cisco 9971 phone configuration working example with setup tips

[tenaro.gusatu.novici]

Hi Leo,

I do appreciate your willingness to help but you are acctually doing completely opposite. I'm writing this only to help other people who will be reading this post. SIP is by definition peer-to-peer protocol (can be easily found in any decent book about VoIP protocols or how you like to say it, google it) meaning it allows communications between two entities (User Agents) without some call-agent or similar call control device. Further, when you talk about codecs, udp/tcp, IP addresses etc it is solved by SDP. Again, nothing really special here, it is well known and well defined common procedure. Depending on SIP EO (early offer) or SIP DO (delayed offer), teminal capabilities are sent using SDP as part of initial INVITE request or in 200 OK response and then challenged against opposite side SDP (that is injected into 200 OK response or ACK request).

[Leo Laohoo]

You still need a call server.  The Cisco phones are DUMB.  These machines need something to talk to.  Something that'll manage the calls from start to finish.

Consider this ... If these phones can make direct calls, whether by SIP or SCCP, do you think Cisco's voice portfolio would be this healthy?  Let's not talk about Cisco, what about other providers like Nortel or Avaya.  Without any call servers, do you think their phones can talk directly to each other without any a call server?

[tenaro.gusatu.novici]

Every physical Tandberg endpoint is able to do direct SIP calls, in case you have smaller deployment and want to do it that way. In case customer is ready to invest into VCS (call control device) then you can register all those endpoints to the VCS.

[Leo Laohoo]

Every physical Tandberg endpoint is able to do direct SIP calls

I'm no expert with Tandberg products but you are talking about Cisco 6K, 8K and 9K products.  And these endpoints are dumb.

Besides, you can't say that Paolo is wrong.  He's been in the voice and routing business for so long he's knows what he's talking about.

[Leo Laohoo]

 So, no VSP, I simply want to understand what needs to be configured to make direct call.

As what Paolo has posted, you need a VSP and a call server.  The call server talks to your VSP (much like a home modem/router talks to your ISP) and your phones talk to your call server (like the relationship with your PCs at home to your home modem/router). 

Google search for the following:  Asterisk, FreePBX, TrixBox.

You can host either one of these either in a cheap Raspberry Pi or Beagle Bone.