rtpproxy.org

RTPProxy

A high-performance software proxy that brings control to your VoIP network. VoIP traverse NAT firewalls. Relaying of voice, video or any RTP stream of data. Playing of pre-encoded in-band announcements. Routing VoIP calls over VPN links. Support for FreeBSD and Linux. A simple control protocol allowing for integration with other systems. BSD Clause-2 licensed code. Clustering across geographic areas (with assistance from rtp cluser). Integration with SER, Kamailio and Sippy b2bua. SIP Express Router (SER).

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Sippy Software, Inc.

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Sippy Software, Inc.

Maksym Sobolyev

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so●●●●●@sippysoft.com

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Sippy Software, Inc.

Maksym Sobolyev

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Bel●●●ham , WA, 98226

US

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so●●●●●@sippysoft.com

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RTPProxy | rtpproxy.org Reviews

<META> DESCRIPTION

A high-performance software proxy that brings control to your VoIP network. VoIP traverse NAT firewalls. Relaying of voice, video or any RTP stream of data. Playing of pre-encoded in-band announcements. Routing VoIP calls over VPN links. Support for FreeBSD and Linux. A simple control protocol allowing for integration with other systems. BSD Clause-2 licensed code. Clustering across geographic areas (with assistance from rtp cluser). Integration with SER, Kamailio and Sippy b2bua. SIP Express Router (SER).

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RTPProxy | rtpproxy.org Reviews

https://rtpproxy.org

A high-performance software proxy that brings control to your VoIP network. VoIP traverse NAT firewalls. Relaying of voice, video or any RTP stream of data. Playing of pre-encoded in-band announcements. Routing VoIP calls over VPN links. Support for FreeBSD and Linux. A simple control protocol allowing for integration with other systems. BSD Clause-2 licensed code. Clustering across geographic areas (with assistance from rtp cluser). Integration with SER, Kamailio and Sippy b2bua. SIP Express Router (SER).

INTERNAL PAGES

rtpproxy.org

1

Backstory

http://www.rtpproxy.org//about

RTPProxy was originally developed by Maxim Sobolyev in 2003 for the purpose of enabling VoIP calls to/from SIP User Agents located behind NAT firewalls. Several case existed where direct end-to-end communication between NATed SIP User Agents was not possible. RTPProxy, in conjunction with a SIP proxy overcomes these impediments by acting as a go between for the RTP streams. Subsequently, the RTPProxy became widely used by VoIP service providers that needed to optimize traffic flow in their networks.

2

Backstory

http://www.rtpproxy.org/about

RTPProxy was originally developed by Maxim Sobolyev in 2003 for the purpose of enabling VoIP calls to/from SIP User Agents located behind NAT firewalls. Several case existed where direct end-to-end communication between NATed SIP User Agents was not possible. RTPProxy, in conjunction with a SIP proxy overcomes these impediments by acting as a go between for the RTP streams. Subsequently, the RTPProxy became widely used by VoIP service providers that needed to optimize traffic flow in their networks.

3

Solaris, improved re-INVITES, port randomness and more.

http://www.rtpproxy.org/post/ability_to_fork

Solaris, improved re-INVITES, port randomness and more. Latest snapshot of development 1.2 version of RTPproxy has been released. This release includes many bug fixes and new features, including:. Support for the new. Improved support for re-INVITE. New Snapshot: Solaris, you compile me.

4

Posts

http://www.rtpproxy.org//post

Rtpproxy v2.0.0 release. Mon, Mar 9, 2015. Reorganization and new developments for RTPProxy. Thu, Jun 19, 2014. Mon, Jan 12, 2009. New Snapshot: Solaris, you compile me. Mon, Dec 8, 2008. Solaris, improved re-INVITES, port randomness and more. Tue, Nov 11, 2008.

5

Beta 1.2

http://www.rtpproxy.org/post/beta_one_point_two

New beta snapshot of 1.2 version of RTPproxy has been released. This snapshot includes;. Improvements in re-packetization feature. New Snapshot: Solaris, you compile me.

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iptel.org

About SIP Express Router | iptel.org

http://www.iptel.org/ser

About SIP Express Router. November 04, 2008 - SER i. Joins the sip-router project. Read more about it in the announcement. SIP Express Router (SER) is a high-performance, configurable, free SIP server licensed under the open-source GNU license . It can act as SIP ( RFC 3261. Registrar, proxy or redirect server. SER can be configured to serve specialized purposes such as load balancing or SIP front-end to application servers, SEMS. For example. Together with the RTPproxy. Load-balancing), NAT i. Tested an...

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Kamailio Behind NAT | Fred Posner

http://www.fredposner.com/1457/kamailio-behind-nat

The World of FISPosner. After returning home from AstriCon 10, I decided to start-up a new server and see how long it would take me to run a working Kamailio server behind NAT (on a private IP). Bottom line? I was lucky to help staff the Kamailio booth this year and was honored to help so many other VoIP professionals (and enthusiasts) with their questions about Kamailio, so without further ado…. WebSocket support for WebRTC. Who should use Kamailio? 8230; but … I digress …. Running Kamailio behind NAT.

saghul.net

MediaProxy | Mi Brain-Training Personal

http://saghul.net/blog/tag/mediaproxy

Para que no se me olviden las cosas…. Archivo de la etiqueta: MediaProxy. SIP2SIP: SIP gratis para las masas. Hace unos pocos días tuve la oportunidad de participar nuevamente en el voip2day. Con la charla “SIP2SIP: SIP gratis para las masas”. El evento fue increíble, sin duda la mejor edición hasta la fecha! Pude reencontrarme con antiguos amigos, hacer nuevos y desvirtualizar a otros cuantos. Muchas gracias a los organizadores y. Por seguir trabajando en hacer un voip2day mejor cada año. En su día escr...

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Related Projects | The Kamailio SIP Server Project

http://www.kamailio.org/w/related-projects

Helpers to build and maintain SIP platforms, developed by members of the project team. Email us at. If you want to add new projects in this page. SIREMIS Web Management Interface. Open source project is an easy to install cross platform web management interface for admins of VoIP platforms based on Kamailio (OpenSER). Project. Details about installation as well as screenshots and a demo are available on the project web page at:. Http:/ siremis.asipto.com. Http:/ www.sipwise.com/products/spce/. WeSIP is a...

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Downloads | iptel.org

http://www.iptel.org/download

Raquo; Using SER. The most recent released version of SER. Is 096, also referred to as 0.9.x. Go to iptel.org's FTP server for download of. Latest stable release 0.9.6. Previous version are also available. Snapshots of upcoming patch release for SER 0.9.x. Are also available. Patch releases contain bug and security fixes. You probably want to have a look at the SER - Getting Started document. If you are going to use SER with RADIUS i. Has been made available (note! For information about this functionality.

ftp.iptel.org

About SIP Express Router | iptel.org

http://ftp.iptel.org/ser

About SIP Express Router. November 04, 2008 - SER i. Joins the sip-router project. Read more about it in the announcement. SIP Express Router (SER) is a high-performance, configurable, free SIP server licensed under the open-source GNU license . It can act as SIP ( RFC 3261. Registrar, proxy or redirect server. SER can be configured to serve specialized purposes such as load balancing or SIP front-end to application servers, SEMS. For example. Together with the RTPproxy. Load-balancing), NAT i. Tested an...

powerpbx.org

Kamailio Supernode & Siremis GUI Install guide. | Asterisk FreeSwitch guides

https://www.powerpbx.org/content/kamailio-supernode-siremis-gui-install-guide-v1

Asterisk and Freeswitch Hosting. Kamailio Supernode and Siremis GUI Install guide. On Sat, 03/21/2015 - 09:26. A Kamailio supernode is a SIP router capable of user authentication and status tracking among other things. It can be used to create a private secure peer-to-peer SIP service similar to Skype. For example. It can also be used to connect to other nodes, gateways, PBX's etc. Siremis. Appears to currently be the best GUI for use with Kamailio. This guide was tested using:. Define WITH MYSQL #!

kb.asipto.com

kamailio:kamailio-mixed-ipv4-ipv6 [Asipto - SIP and VoIP Knowledge Base Site]

http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6

Asipto - SIP and VoIP Knowledge Base Site. Book: SIP Routing With Kamailio. Learn how to build your own real time communication service! Click Here for More Details! Run your own SIP VoIP service on both IPv4 and IPv6. Run your own SIP VoIP service on both IPv4 and IPv6. Main author: Daniel-Constantin Mierla miconda [at] gmail.com - founder Kamailio SIP Server project. At the date of publishing the article, the world celebrated IPv6 day. Networks using open source Kamailio SIP server. Did I say NAT?

saevolgo.blogspot.com

SaevolGo: How To: Increasing VoIP Services Capacity

http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html

Just some VoIP Stuff I Learn from internet and return the knowledge back to the internet. Friday, November 11, 2011. How To: Increasing VoIP Services Capacity. This is the second part on increasing voip services capacity. In the previous post. I had a high level overview of what an SBC is and how to radically increase the call-capacity. In this post we'll proceed with the architecture setup and configurational steps required. So Install a Ubuntu Server and then follow this blog post. Above blog post is a...

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Liberty's VX and VXA RTP Goodness | A collection of old and new RTP edits… and a few originals here and there~

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Ride to Pike's Peak Route Detail. In 1806, Zebulon Pike. Traveled west and sighted Pikes Peak. We biked 990 miles along his route and then ran the Pikes Peak Marathon. View my complete profile. Main Ride to Pike's Peak Blog Journal. August 3rd to 6th. August 7th to 9th. August 10th to 12th. August 13th to 15th. August 16th to 18th. August 19th to 20th. Ride Along With Us. Follow this link for a PDF. Showing the final mileage and stops for the route. Posted by - 3:14 PM. Posted by - 10:14 AM. August 3rd w...

rtpproxy.org

RTPProxy

A high-performance software proxy that brings control to your VoIP network. VoIP traverse NAT firewalls. Relaying of voice, video or any RTP stream of data. Playing of pre-encoded in-band announcements. Routing VoIP calls over VPN links. Support for FreeBSD and Linux. A simple control protocol allowing for integration with other systems. BSD Clause-2 licensed code. Clustering across geographic areas (with assistance from rtp cluser). Integration with SER, Kamailio and Sippy b2bua. SIP Express Router (SER).

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