18 years ago
I think it would be easier to help you if you provided more information about what you are trying to achieve. Are you trying to set up a certain piece of equiment? If so, what kind?
Anyone can explain me please, or tell me if exist a link, for explanation about VOIP paramters:
DTMF: inbound outbound payload type sip info
Please, a good link just for learn about VOIP Thank you
Doesn't talk about inbound-outbound and others parameters... Btw thanx
Typing "DTMF inbound outbound payload type sip info" into Google gets a bunch of hits (of course). One of the ones midway down is
Hope this helps.
I have two draytek VigorTalk. They works also on a peer to peer instance, but i need to know and learn more about DTMF paraeters, no manuals on the website about it. That's all. Thanx in advance.
Usually, these parameters refer to how you'd like to pass DTMF digits to the other end of your call. H323 or SIP signaling will negotiate your choices with the other end to help establish call parameters. The choices can be set up for what you want to send (outbound) and what you want to receive (inbound). Choices are generally:1) In band digits: digits are passed along with the rest of your voice, using the same VoIP codec as your voice (G.711, G.729, etc.)
2) Out of band digits using special IP pakets that are sent along with the voice path. These packets use RFC 2833 format, these packets must have a "payload type" that matches what the other end is listening for. Cisco often defaults to 101 for this payload type, which uses your definition to negotiate (again via H323 or SIP) with the other end during call establishment. These digits are pulled out of the voice stream and sent only in the RFC 2833 packets.3) Out of band digits sent via the signaling layer (UII packets). These have no payload type defined. The digit is removed from the voice path and signaling messages are sent to tell the other end about the digits.
In 2 & 3, when the other end receives these special packets that represent digit value & duration, it will regenerate the proper tone and duration back to the voice path.
The purpose of this is to guarantee a valid tone detection at the distant end of the call, since some voice codecs may not be able to do that properly (other then G.711 it's a gamble).
Hope this helps,
Greg > "Miguel Cruz" wrote:
Greg.... THANK YOU VERY MUCH! Really, clear and comprensive, thanx a lot friend. :-)
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