Upgrade 3CX to v18 and get it hosted free!
Home | SIP | SIP Outbound

SIP Outbound

Author image
Posted by , on January 25, 2013

SIP Outbound is a SIP extension that focuses on keeping SIP connections alive over a NAT. The SIP user agent sets up two TCP/IP flows over UDP or TCP to two separate SIP proxys, called Edge Proxies. If one connection dies, the other takes over. The service user these connections for requests from the core to the user agent. All responsibility for NAT traversal is pushed to the edge, to the SIP UA in SIP outbound.

Implementations

  • Kamailio 4.0 includes SIP outbound Support
  • Counterpath Bria for iPhone supports Outbound
  • BareSIP

Article Reviews

Write a Review

Your email address will not be published. Required fields are marked *

Required Field. Minimum 5 characters.

Required Field. Minimum 5 characters, maximum 50.

Required field.There is an error with this field.

Required Field.

This site uses Akismet to reduce spam. Learn how your comment data is processed.

There are no reviews for this article. Be the first one to write a review.

Related Posts:

SIP Trunking vs PRI
December 2, 2021
Have you converted to Voice over Internet Protocol yet? If so, you are probably looking for a way to integrate VoIP technology with traditional analog phone lin...
PortSIP Softphone App provides push to talk
December 6, 2019
PortSIP Softphone is the industry’s most reliable and feature-rich communication application for iOS, Android and Windows. Based on SIP Standards, it becomes ...
SIPphone
August 2, 2005
Website: SIPphone.com Service launched in 2004 by the founder of Lindows and MP3.com. Offers many unique and interesting features including a do-it-yourself con...
Mitel MiNET to SIP Conversion
May 1, 2007
The typical Mitel phone ships from the vendor in MiNET mode and the firmware version present will most likely not support SIP mode. This guide explains how to c...
Asterisk sip allowguest
July 15, 2006
sip.conf: allowguest This setting determines if anonymous callers are permitted to place calls to Asterisk. This defaults to ‘yes’. You will need to...
SIP method cancel
September 11, 2003
From the SIP RFC: The CANCEL request, as the name implies, is used to cancel a previous request sent by a client. Specifically, it asks the UAS to cease process...
Sipura Technology Inc.
September 3, 2005
Sipura was purchased by Linksys and the Sipura product line has been rebranded to Linksys Update Jan 2009: Linksys Voice products (what was Sipura) are now goin...
Asterisk sip directrtpsetup
September 13, 2008
Version Introduced in Asterisk 1.4 Description directrtpsetup=yes is similar to directmedia=, except the audio is redirected in the initial INVITEs rather than ...
sipX Solution Summary
May 13, 2005
sipX – Easy to use, scalable, and feature rich open source unified communications and IP PBX solution for Linux SIPfoundry info: Original Website: http://...
Centile
May 12, 2004
https://www.centile.com From website: Founded in 1998, Centile Telecom Applications is the leading European developer of unified communications platforms for op...
Get 3CX - Absolutely Free!
Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Verify your Email

Check your inbox!

We’ve sent you an email. Click on the button in the email body to verify your email address – (if you can not find it, check your spam folder).

Upon verification you will be directed to the 3CX setup wizard.