Basic Question IPT

[mahmoodmkl]

Dear Gents,

Please excuse me for asking such a dumb question as i am trying understand the technology.Please could someone explain.

1.How IP Phones work i.e how the calls work.

2.How spoken voice becomes the pay load of RTP.

Regarding  the first point i think the call manager is responsible for call setup  and tear down.I mean the call manager will provide the necessary  information to the IP Phones.

Regarding  the second point i m very confused.I understand that there are codecs  involved in this function but i need to understand how is the step by  step  process from  ground level.

I will come up with some more questions once i recieve the answers for the above.

Thanks

[Hermanus Janse van Vuuren]

Read this

http://www.net130.com/tutorial/cisco-pdf/Cisco_%20IP%20Telephony%20Network%20Design_Guide.pdf


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[Suresh Subramanian]

The built-in codecs (COder+DECoder) of the IP phones perform the digitization, compression and packetization process            directly within the phone and send the resulting stream of packets over            an Ethernet connection.

URL: http://www.voipsupply.com/voip-phone

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[mahmoodmkl]

Hi,

Please can u explain where these codec are present i.e in software or hardware.

Thanks

[Suresh Subramanian]

The receiver mic coverts the speech to analog audio singal and the builtin codec coverts it into digital signal. DSP is the chipset inside the ip phone.

http://en.wikipedia.org/wiki/VoIP_phone#Hardware

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[Jaime Valencia]

If you seriously want to understand how this works, read this, and further reading in the nyquist theorem

http://en.wikipedia.org/wiki/Digital_signal_processing

http://en.wikipedia.org/wiki/Analog_signal_processing

http://en.wikipedia.org/wiki/Digital_signal_processor

There's a DSP in the phone that does all this, just as there is in pretty much anything you know that can talk, toys, cars, mobiles, etc. They turn digital info from their memory bank, into an analog signal that outputs thru the speaker.

HTH

java

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[mahmoodmkl]

Hi,

Thanks for your replies.

So you mean that when you pick up the handset and start talking then the DSP will start their job.

In my understanding the DSP are capable of converting  our voice in different codec formats i.e either g.711 or g.728 etc.

Then  the coded data is coverted into bits wrapped in RTP/UD/IP and sent to the other phone..am i right in understanding this.

The codec which needs to be used by the IP Phones is decided by the Call manager.

Once the call manager provides the necessary information to both the IP Phones it gets out of the picture.

Then the data flows directly between the IP Phones...is this correct in my understanding...?

Thanks

[mahmoodmkl]

Hi,

Anybody can help.

Thanks

[Terry Cheema]

Yes thats right. RTP audio flows directly between end points. CUCM still controls signalling - for providing any mid call features like if you put a call on hold, transfer, conference, etc. it can still provide you those features.

DSP will come into picture when one leg of call is TDM and will do voice termination and conversion. Basically analog to digital (study Nyquist theorem if you want to know how its done).
Apart from that DSP also provide transcoding, conferencing etc. DSPs are actually chips on a PVDM card which looks similar to RAM and goes into voice gateway router.

Hopefully clears your confusions.

-Terry




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[Manish Gogna]

Hi Mahmood,

The codec conversion capability ( like between g711 and g729 ) is not possible with the dsp's available on the IP phone. For such functionality we need to configure transcoder on a gateway and it uses the DSP's available on the gateway. You are correct about the following part 'The codec which needs to be used by the IP Phones is decided by the Call manager. Once the call manager provides the necessary information to both the IP Phones it gets out of the picture. Then the data flows directly between the IP Phones...is this correct in my understanding...?'

Once the call is establishe RTP flows directly between the endpoints.

HTH

Manish

[mahmoodmkl]

Hi Manish,

Thanks for your reply.

Actually i phrased my question in a wrong way.I understand the xcoding cannot be done by IP Phone.

My intention was that the IP Phone will  convert the voice using the built in DSP as per the codec which is specified by the CUCM.

i am confused about the formation of packet and conversion of voice from analog to digital and to payload..could you please explain..?.

I understand that DSP's are responsible for the converstion of voice from analog to digital,i want to know what happens after the converstion i.e how is the ip packet formed with the payload.

Thanks

[mahmoodmkl]

Hi,

Could anyone explain the above.

Thanks

[Manish Gogna]

Hi Mahmood,

Please check the section "Major Steps of Voice Processing in VoIP" of the following link

http://www.ciscopress.com/articles/article.asp?p=1687880

I couldn't find a more granular document. Let me know if it helps.

Manish