All posts under tagged ‘SIP proxy’

Feed for all posts filed under "SIP proxy"

Free Calls With SIP Proxies, Asterisk, And A Bit of FreePBX.

Source: snapvoip.blogspot.com

Connect for free, talk forever, pay nothing.
Nerd Vittles (NV) always surprise me one way or another. Today it is SIP Proxies. Last time I looked into Asterisk SIP proxy was when a client needed me to consult on firewalled Asterisk. As the link suggests, I used SIPproxD for that purpose. I think it is still a good solution for firewalling or masquerading an Asterisk Server.
Then there is Dundi, (if you do not know what Dundi is read the Asterisk Documentation or wait till NV writes about it, they promised.
It was easy for me to test the NV demo as were were testing PBX in A Flash both here in USA and a far north corner in Japan. We skipped the kick-ass.net and configured our own DNS servers. Followed the instructions and it worked. (Had to correct my own typos a few times, typing English on a Japanese keyboard is always hard)
But as NV mentioned, what amazed us was the difference in call quality. Now we will have a permanent SIP proxy serving Japan and USA.
In addition, we now have IPKall numbers and we know the phones we use has been selected by NV to be the best IP Phone, the Aastra 57i CT.

Published on March 14th, 2008 under ,

Cisco uses OpenSER, an Open Source SIP router and more in Linksys One.

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com
It is coming in to light that big name companies are using more and more Open Source products. It is no difference when it comes to VoIP. I remember a bunch of companies that used OpenH323 before SIP came to be the leading protocol.
It is revealed that Cisco is using OpenSER as SIP proxy for Service node in Linksys one communication platform.
OpenSER is not the only Open Source Package in the Service Node;
The Cisco Service Node servers run a collection of open-source and Linksys One software:

• FreeBSD-This is the open-source operating system that runs on all Cisco Service Node servers. FreeBSD provides a mechanism that allows multiple virtual instances of the OS to be spawned and run on the same server, with each virtual OS completely isolated from all other instances. This is the partitioning mechanism used to implement the brand-level services.

• PostgreSQL-This open-source package is used to provide database services on the Cisco Service Node.

• OpenSER-This open-source package is used as the Cisco Service Node SIP proxy.

• BIND-This open-source package is used for Domain Name System {DNS) services. The Cisco Service Node runs its own DNS servers. DNS is used for several functions on the Cisco Service Node, including ENUM-based call routing of SIP calls and branding (each brand is known to the outside world as a separate DNS domain name).

• BIND DLZ-This open-source package allows BIND to use the PostgreSQL database to store its zone information. Dynamically loadable zones (DLZ) allows DNS updates to be reflected immediately when a change is made to zone data in the database. This feature is important because CPE that uses Dynamic Host Configuration Protocol (DHCP) can change its IP address at any time. When this happens, DNS must be updated immediately for the ENUM-based call routing to be able to successfully route calls to the CPE.

• NET-SNMP-This open-source SNMP package runs as an agent on the servers and implements several MIBs.
Thank you Cisco for letting us know so openly ;) and congrats to OpenSER people for providing such quality product.

Links;
The OpenSER mail archive entry and discussion


OpenSER gets CISCOs vote of confidence

Cisco Service Node for Linksys One SN-10 and SN-100 Data Sheet

Published on March 11th, 2007 under , , , , , , , , , , ,

What SER is and isn’t

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com

Sys Admins view of SER guts!
The following information is from SER site news report, which in turn plucked from a discussion on a Developers mailing list. It is so important so that I have entirely reproduced the post, for my own reference.
"Consider a more simple SIP proxy like repro. All you can do there is start the damn thing and give it the user data (what would be subscribers, aliases, and parts of the usr_preferences in SER 0.9). Sounds all nice and simple.

Now, as an VoIP operator, my world will be a little bit more complicated. I may have different services that run on separate proxy farms. I may have interesting add-on services like call forwarding, voicemail, IVRs, whatever else product management comes up with. Somewhere in a dark corner, I have some PSTN gateways or, instead, I have an agreement with some telco to do that for me.

If you draw this, you’ll get at least half a dozen boxes with weird connections between. If this doesn’t scare you, start sketching the call flows. You will suddenly find little funny quirks, that of course you can put into C code but if why? SER provides you with the opportunity to solve pretty much all of them in a very simple language.

Better yet: You write your script, you do a test call. If it doesn’t work, you make a trace, you fix your script and try again. No compiling, no packaging, just a restart (BTW, something for the wish list: reloading the config on a SIGHUP). Another trace, another round.

Now we fast forward a bit. Your system is running just fine. But one of your PSTN interconnect partners updates their software and — surprise — all the calls to them fail. Sure, you could use another partner. But your friends in billing will tell yet that their prices for some destinations are just insane. We _really_ have to have that first partner.

Sure, you quickly figure out what the problem is. Sure, you call them and try to explain to the unfortunate fellow on the other end how SIP works and why their stuff isn’t really SIP. Sure, after a while they give in and promise to fix it. But can they do that quickly? Nope. They have to go and talk to whoever delivers their software.

Half a year passes and nothing much happens.

Now, with SER all I need to do is find the route for the specific partner, do the magic with subst() and maybe some other horrible things and voila, it works. Everyone is happy. And should the partner actually ever get their stuff fixed, I can just remove those three lines I had to add.

With repro, things would have been quite different. I have to know enough C++ to actually grok their design or have to have someone doing this. Implementing the three line fix, testing it, producing it easily takes a man-day. With SER I did that in three minutes. Including
the test call.

What it comes down to is, that there is no universal thing. For NATi, there isn’t six funny devices that you find a work around, report to the good folks at iptel, who then add another flag. NAT routers change with every software revision. Old things go away, new things pop up. It is your responsibility as a provider to stay close. That’s what people pay you good money for.

Sure, SER is hard to get into as a beginner. If you want to stay a beginner and don’t care about SIP, use repro. It’ll probably work for you out of the box. If you expect to have to do more, invest the time, learn SIP, learn the ser.cfg. It will pay off later. Everything will be "SER gut" (Sorry, that just had to happen)."

SER Home

Published on February 17th, 2007 under , , , , , , , , ,

Load Balancer for Asterisk and VoIP, from VocalScape

Source: snapvoip.blogspot.com

VoIP IP Telephony, snapvoip.blogspot.com.
Via PRNEWS WIRE/yahoo

VocalScape Networks announced today that they have released a load balancer for VoIP IP Telephony usage.
The Vocalscape Load Balancer began as an open source project which was adopted and improved upon by Vocalscape. It was made compliant with Asterisk, a popular open source PBX, and the algorithm was revised to more evenly distribute calls. Previously, the Load Balancer would send calls to a primary server and only when the primary server was overburdened would calls be sent to additional servers. The new algorithm balances the load by evenly distributing the calls between the servers. As an additional benefit, the Load Balancer provides failover capabilities. If a server is not responding, the Load Balancer will route all calls to servers that are functional.

"Vocalscape has developed the Load Balancer to meet our customers’ needs," commented Ron McIntyre, President of Vocalscape. "As our customers grow their user base, they will need to add additional servers to handle the higher volume of calls. The Vocalscape Load Balancer will allow them to evenly share the load among multiple servers."

The earliest known (to me) SIP load balancer was at Vovida.org. May be this is the one they improved. Vovida Org was one of the pioneering VoIP opensource sites that became less functional. Follow the links to get an idea of how it was like, to develop and write VoiP applications in those days (Ha! it was only 5-6 years ago).

Links;
Vocalscape
SIP Load Balancer at Vovida.org
Yahoo news

Open source SIP stack released

Source: snapvoip.blogspot.com

Open source SIP stack OpenSIPStack has been rereleased under a triple licensing scheme to ensure that it can be used by the largest possible number of individuals and development communities. This tri-license aims to address the perceived incompatibilities between Mozilla Public License (MPL), GNU General public license (GPL) and GNU Lesser Public License (LGPL). The stack was previously distributed under MPL 1.0.

Open Source SIP project, openSBC is based on the OpenSIPStack, a fully compliant (RFC 3261) SIP stack designed for stability and scalability, and with a heritage of commercial usage. The project currently contains reference implementations of a session border controller (OpenSBC), Yeya and several components that are useful to developers wishing to use Solegy’s service deployment platform.
OpenSBC

OpenSBC is a reference implementation of a hybrid SIP proxy and B2BUA (back to back user agent) created from the Open SIP Stack core. It is well suited for a number of VoIP implementations. Among other things, it can be used as a Registrar for SIP endpoints, as an entry/egress point for SIP trunking applications, or as a far-end NAT traversal solution.

OpenSBC has been designed for scalability and flexibility. Deployments can grow incrementally with traffic needs because a primary instance can be configured to load balnce sessions across other instances. Each instance may be run on separate servers, or multiple instances may be run on a single server.

OpenSBC can perform the following functions:

Session Border Controller: Full back-to-back user agent (B2BUA) hides network topology with:
- Integrated web UI for basic configuration tasks
- far-end NAT traversal with RTP proxy
- Complete transparency for end-nodes with support for pass-thru of non-standard SDPs,
- Routing using static rules, ENUM or Solegy RTBE
- Comprehensive logs using syslog server
- Encryption of SIP and RTP packets with simple hash

Registrar: Fully standards-compliant with support for pass-through registrations (also referred to as upper registration) and integrated support for presence using SIP/SIMPLE or XMPP.

Proxy: Fully standard-compliant with multi-protocol support (UDP, TCP, TLS*), processing and relaying signals from remote (SIP) and local endpoints.

Presence: Compliant with SIP/SIMPLE and XMPP standards with support for PUBLISH as well as SUBSCRIBE/NOTIFY events.

Event Packages: Support for message-summary information about waiting messages (voicemail) and presence

Solegy™ Offers Free VoIP Softphone for Microsoft Windows — Customized Softphones Available to Service Providers with a Full Range of Calling Features and Back-Office Functionality in a Hosted Environment.

Links;
Open Source Sip website
OpenSIPStack web site
Solegy website

Published on December 19th, 2006 under , , , , , , , , ,

Member of "Hype Media! Network"