within the internet clouds, many protocols can be used to pass the signaling
information, eg. SIP, H323 or
even ISUP/TCP. Refer to SIP RFC or H323 spec. for details.
thanks,
John
-----Original Message-----
From: Alok Dubey (OCS-BLRAKS-AVS) [mailto:adubey@wipro.co.in]
Sent: Wednesday, February 21, 2001 3:12 PM
To: 'Li, John '
Cc: 'ITU-SG16(a)mailbag.cps.INTEL.COM'
Subject: RE: a query
hi,
I agree it goes via SG to MGC..but that is when im going acros
medias....POTS to VOIP etc.. but within the IP cloud.. how does the
signalling take place.. eg from VOIP to VOIP..
lets say i have 1 MGC (or u can assume this to be an IP phone too) A and
another ip phone B in a diff LAN seg ( or a POTs phone attachd to a
different gateway...)
now when i call from A( if its an MGC ill use A as a gateway) to B , A uses
the gatekeeper/callmanager to get the number (ARQ) and then depending on the
mode of call.. . it sets up the call from A to B by either going via the
gateway or by seting up the session directly..
now this part of the signalling is what i am interested in.. is it ordinary
TCP/IP with some good old ports or is it RTP too?
if it isnt.. as u put it .. how do i make my network elements mark/identify
and prioritise this traffic. will i have to do it at the ingress node.ie
when it enters the network?
and what is the content of the packet? ..is it well defined?
Rgds
Alok
-----Original Message-----
From: Li, John
To: 'Alok Dubey (OCS-BLRAKS-AVS)'
Sent: 2/22/01 1:08 AM
Subject: RE: a query
Alok,
The call control messages (including setup) usually goes thru the
signaling
path via SG and MGC.
RTP is used for the media tranfer.
Hope it helps.
John
-----Original Message-----
From: Alok Dubey (OCS-BLRAKS-AVS) [mailto:adubey@WIPRO.CO.IN]
Sent: Wednesday, February 21, 2001 2:13 PM
To: ITU-SG16(a)mailbag.cps.INTEL.COM
Subject: a query
hi
im sorry for posing this query out here but i needed this answer asap..
Im basically more of an implementor than a protocol developer and all my
job
is above the layer 3.. ie socket programming etc.
Right now while going thru the specs, I could not find any reasonable
explanation on how u prioritise traffic for call setup?
for eg are u sending callsetup info over RTP ports too..? if yes, i
still
have to figure out how to use Voice Activity Detection to conserve
bandwidth
a feature which lets us drop voice packets below a certain db level ..,
if
no this basically means that setting up a call could take a long time
and
the traffic associated with it cannot be controlled/prioritised..
can anyone refer me to any good specs
Sorry to be wasting ur time on such a query.. but i couldnt think of a
better place to pose this question.
Rgds
Alok
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