SIP deployment scenarios questions

[sarahr202]

Hi everybody

I have few questions about the possible SIP scenarios; I just want to confirm if I am correct.

Scenario#1

CISCO IPphone---SIP--Call manger

Can we use SIP signalling between IP phone and call manger instead of cisco's SCCP?

Scenario#2

IPphone--SCCP---Call manger---SIP-----gateaway--PSTN-------PSTN_PHONE.

Can we use SIP between Call manger and gateway?  this is my understanding:

Ip phone registers with Call manger using SCCP, Let suppose Ipphone wants to call PSTN phone, Call manager received signalling info from IPPHONE via SCCP such as called number, ,call manger based on the Dial plan, then concludes it has to contact gateway using sip signalling., 

after all signalling is done, RTP stream flow directly between IPPHONE and gateway.

Scenario#3

IPhone-SCCP---Call manger---SIP-------SBC-------PSTN--PSTN_PHONE.

Is it possible ?

Much appreciated!!

have a great weekend.

[Terry Cheema · Accepted solution]

1) Yes absolutely you can and the latest 7800/8800 models are infact SIP only.

2) Yes you can use SIP between CUCM and GW and again is getting popular with the upstream PSTN connection through SIP trunk to provider instead of traditional ISDN lines. Yes RTP flows directly between end points. CUCM only handles the signalling.

3) Yes possible no issues at all.

Let me know if you have any more questions.

-Terry

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[sarahr202]

thanks Terry,   I will have one more question ; since it is totally different issue, ( TURN: Traversal Using Relay NAT ), I will post it separately.

[John Cheetley]

Hi Terry,

Following on from your responses to sarahr202.

Is a SIP server (or proxy) required for each end point for SIP to function either for outgoing or incoming calls

Is a SIP server (or proxy) required for outgoing to ISP then to mobile or PSTN?

Do you need a router on each end for VOIP calls via SIP to successfully connect?

Thanks in advance

[Terry Cheema]

John,

SIP is based on client-server model. End points are clients and they need a server to function. In the above enquiry CUCM acts as server (Registrar/B2BUA), you can have any other server like astresik etc.

For PSTN calls there are normally three components involved:

End Point (Your Cisco Phone) > Server (In SIP context this may be more than server or one server hosting multiple roles, CUCM for example can act as registrar and B2BUA)>SBC(Session Border Controller - CUBE Router for example)>Service Provider

SBC is there for security/capabilities matching etc. SBC can be a router or appliance like Acme Packet. You can have direct SIP Trunk from your server to the Service Provider as well (but not recommended) for PSTN calls.

Let me know if you have further questions.

-Terry

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[John Cheetley]

Thanks Terry for your reply.

My SIP service is active as the router advises. How do I get the SIP phone to register please as when i run show sip-ua register service, system comes back with no registered services but gives me line 0300 and 1002. I have attached the output if that helps

cme_router#sh sip-ua register status
Line          peer           expires(sec)  registered
============  =============  ============  ===========
0300          20001            29            no

However. When i do sh sip-ua statistics.

shows registered 401 times and wireshark shows SIP transactions being processed

And SIP service is Enabled

Thanks in advance.

Sorry for the late reply. 

[John Cheetley]

Evening Terry,

Sorry but I probably overwhelmed you with some information before.

Just wanted to know if a sip server is mandatory for each endpoint. I know ISP's use them but a SOHO for 2-3 phones environment.

And are ephone dn and DID's (direct inward dialling) the same?

I understand that ethernet phone is a directory number in CME representing the "line" that connects a voice channel to the voice port but a DID is what incoming calls come in on via a dial peer.

Isn't that the same thing?