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mahmoodmkl
Rising star

Call Flow between SIP/SCCP phone which is registered in CUCM with H.323 Gateway

Hi,

 

Please can someone clarify how is the call flow taken place between an IP Phone using SCCP/SIP with H.323 gateway.

What is the role of the call manager between this communication.

 

Thanks

 

 

9 REPLIES 9
Manish Gogna
Cisco Employee

Hi,

 

The above call and signaling flow is for SCCP endpoint. In case of SIP phone the communication between CUCM and IP phone would be SIP.

Manish

- Do rate helpful posts -

 

 

 

Hi Manish,

 

Thanks for the reply.

If i understand correctly the CUCM is doing a sort of protocol conversion between the IP Phone and the GW until the RTP is established.

Then the RTP packets are converted to TDM.

Thanks

Hi,

Generally speaking that is true, , cucm goes out of picture once RTP is established.

 

Manish

Hi Manish,

Thanks for the reply.

This leads me to ask you one more question,which is very annoying.

Why are the VOIP protocols implemented,was this not possible with normal IP..?

Is the VOIP protocols implemented just to signalling purpose..?

At the end RTP is also data traffic.

Sorry for the above as i am trying to figure out and understand VOIP.

 

Thanks

 

RTP is just one part of the call, there are other factors like media negotiation ( codec and port ) plus dtmf capabilities. In addition to this there are supplementary functions like hold, transfer , forwarding etc which require allocation of media resources. These functions require a call control and media resource allocation mechanism which would be difficult to accomplish on majority of standalone endpoints.

 

Manish

Hi,

 

Any call is divided into type basic parts:

 

1. Signaling - This is pre-audio phase which includes entering digits, signaling digits, alerting, ringing, etc. For this phase you need a protocol which handle this communication. Here you will have protocols such as SIP, H323, MGCP, SCCP.

2. Media Stream - This is the audio phase where both parties are talking. In IP networks, RTP is the protocol used to carry the actual audio.

 

Hope this is clear now.

Hi,

Thanks for the reply.

So,I assume again these signalling protocols will utilize the underlying IP for all these functions.

 

Thanks

That is correct and this is the power of the VoIP protocols compared to old analog ones

avinsrid89
Beginner

Hello,

Explaining signalling for different call flows is not very easy over a post.

If you give a sample call, it can be explained through traces.

Best way to learn it is to take detailed CCM Service traces from Call Manager and parse it using the Translator X software.

You can see your call flow end to end in detail.

Please rate useful posts and mark them resolved when needed.

~Avinash

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