rfc2833

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Three modes of DTMF (Sipinfo,rfc2833,inband)

1, DTMF (dual-tone multi-frequency) Definition: By the high-frequency and low-frequency sound two sine wave synthesis represents the digital keys (0~9 * # A B C D).2. Methods for detecting DTMF data in sip: Sipinfo, RFC2833, Inband1) SipinfoFor out-of-band detection, DTMF data is transmitted through the SIP signaling channel. There is no uniform implementation standard, and the DTMF keys are identified by the signal field in the Sipinfo package with t

DTMF encoding transmitted by rfc2833 in Voice Transmission

DTMF encoding transmitted by rfc2833 in Voice Transmission 2007-03-23 11:13:48 The class is defined as follows: # If! Defined (afx_head_2833_h000093b5c358_6f19_475b_a49f_13bfcc9dfe1c00000000ded _)# DefineAfx_head_2833_h000093b5c358_6f19_475b_a49f_13bfcc9dfe1c00000000ded _ # If _ msc_ver> 1000# Pragma once# Endif // _ msc_ver> 1000 Class chead_2833{Public:Bool dispelhead (char *Cvalue );Int btodd (char * cvalue );Char * dtob (intNvalue );Che

VoIP DTMF notes

DTMF definition: Digital keys (0 ~ 9 * # a B C D ). There are usually three methods for detecting DTMF in VoIP: SIP info, inband, and out band (rfc2833). In addition, the latest RFC has been adopted for the requirements of DTMF In the 3GPP IMS specification.4733 replaces RFC 2833. 1. Sip info For out-of-band detection, DTMF Data is transmitted through the sip signaling channel. There is no unified implementation standard, which is sent through the S

sip.conf configuration Details

[2001] Type=friend context=localsets host=dynamic Nat=yes Canreinvite=no secret=123456 dfmfmode=rfc2833 Disallow=all Allow=ulaw Allow=alaw allow=h263 Description (1) the type of TYPE:SIP. Format: Type =user|peer|friend. peer is used to authenticate outbound calls, and if you want to have multiple phones in a user (extension), define a extension that can call two SIP peers. User is used to authenticate incoming calls and users reach the server through

Analysis of the principle of DTMF

the loss of DTMF signal due to the influence of network drops, and the DTMF sound is mixed in the voice packet, which is prone to deviation and distortion of signal.C. Transfer via rfc2833 rules and format packageFor in-band detection mode, through RTP transmission, the special Rtppayloadtype is teleponeevent to mark the RFC2833 packet. The same DTMF key will usually correspond to multiple RTP packets, the

Install Asterisk without hardware on Linux

/pseudo" KERNEL = "zap [0-9] *", NAME = "zap/% n" 4. Download Asterisk and Zaptel Cd/usr/src Export CVSROOT =: pserver: anoncvs@cvs.digium.com:/usr/cvsroot Cvs login (the password is anoncvs) Cvs checkout zaptel asterisk 5. Install Asterisk and Zaptel Cd/usr/src/zaptel Make clean Make linux26 Make install Cd/usr/src/asterisk Make clean Make install Make samples Modprobe zaptel 6. Modify some configuration files. Sip. conf and extensions. conf must be modified. Add the following content to the

Rtp sip configuration details

Configuration In the/etc/asterisk/sip. conf file: [General] Context = default Srvlookup = yet; Establish a logic and DNS address method. You can achieve this address and obtain Many DNS benefits. [10000] Username = 10000; User Name Type = friend; user \ peer \ friend can be defined) Secret = 123456; Authentication Password Record_out = Always Record_in = Never; call recording Callgroup; call group. The default value is "1" Pickupgroup; Generation Group Disallow; encod

Basic settings of SIP Trunk in trixbox

Basic settings of SIP Trunk in trixbox Http://www.voclub.net/zone? Action-viewthread-tid-1065The basic settings of the SIP Trunk in trixbox are as follows: the extension can call the phone number through a SIP Trunk, and then ring the trunk number to the extension. Create a new SIP Trunk, provided that you have obtained a sip account that can be connected to an external line. Dial rules: X. Here I set the simplest X. Because X represents any number 0-9, and X represents any length, so this trun

Install asterisk with no hardware on Linux

installCd/usr/src/asteriskMake cleanMake installMake samplesModprobe Zaptel6. Modify some configuration files. There are sip.conf and extensions.conf that need to be modified.sip.conf Add the following, where 1498 and 1499 are my preset two phone numbers:[1498]Type=friendusername=1498Host=dynamicCCanreinvite=yesdtmfmode=rfc2833[1499]Type=friendusername=1499Host=dynamicCCanreinvite=yesdtmfmode=rfc2833Extensions.conf's default paragraph is modified as

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