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SX20 SIP without CUCM, VCS, TMS

Shams Rishad
Level 1
Level 1

Hi

 

Previously I was deploying an SX20 and an SX10 both as standalone and I was told from the community that H.323 is the way to go, not SIP.

 

Now, recently I have deployed an SX20 at another client's premises, this one is standalone deployment as well and I have configured H.323 not SIP, just like I did with the previous one, and calls are being established just fine between H.323 devices.

 

Now client wants us to configure it with SIP, because the other endpoints he is trying to communicate with all are using SIP. Point to be noted, this is standalone, that is to say no VCS, TMS, CUCM or anything like that. Is this possible to make a standalone configuration for SIP? If so, then how?

 

Thanks

Rishad

5 Replies 5

Tommer Catlin
VIP Alumni
VIP Alumni

You have to use a SIP UA server such as VCS or CUCM.    The SIP endpoint has to register to use SIP.  VCS/CUCM will then negotiate the protocols to connect the call.    Really is no point for deploying stand alone video endpoints without call control.  Opening up ports and NAT for IP calling is a huge security risk.

Tommer Catlin
VIP Alumni
VIP Alumni

You will need a server to do this such as VCS or CUCM.  SIP requires a SIP registrar to initiate a call to another device.  H323 and SIP are difference protocols and VCS can make this happen. (or CUCM).

Hi

 

Yes I understand that SIP and H.323 are different protocols, and to make calls form SIP to H323 or vice versa, we will need something like VCS or TMS.

 

But I'm talking about standalone deployment. If both endpoints want to use SIP, is it possible make it happen without CUCM/VCS/TMS as standalone deployment?

 

Rishad

Magnus Ohm
Cisco Employee
Cisco Employee

I agree with Tommer that you should use a registrar. But if all hope is out you can enable a setting that will allow SIP calls without a registrar (IP dialling). You have to enable SIP Listenport in the configurations. So it is *possible* to do this but hardly userfriendly. 

 

/Magnus

Hello Magnus

 

Could you please provide me link to some resource or some guideline to the standalone SIP configuration? Also could you please inform me the possible drawbacks of this configuration? You already mentioned one- not user friendly...

 

Rishad