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@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@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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