Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. If you'd prefer to explore, try our test area to get started. And see here for current known issues.

Basic Question IPT

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

12 REPLIES

Re: Basic Question IPT

Read this

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


Sent from Cisco Technical Support iPhone App

Voice CCIE #37771

Basic Question IPT

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

Please rate all the useful posts

//Suresh Please rate all the useful posts.

Basic Question IPT

Hi,

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

Thanks

Basic Question IPT

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

Please rate all the useful posts

//Suresh Please rate all the useful posts.
Cisco Employee

Basic Question IPT

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

if this helps, please rate

www.cisco.com/go/pdihelpdesk

HTH

java

if this helps, please rate

www.cisco.com/go/pdi

Basic Question IPT

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

Basic Question IPT

Hi,

Anybody can help.

Thanks

Re: Basic Question IPT

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




Sent from Cisco Technical Support iPhone App

Cisco Employee

Basic Question IPT

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

Basic Question IPT

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

Basic Question IPT

Hi,

Could anyone explain the above.

Thanks

Cisco Employee

Basic Question IPT

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

787
Views
13
Helpful
12
Replies