Setting Max Concurrent Call Limit on Kamailio

In this post I'll introduce "Setting Max Call Limit for a Customer Number or IP Address" to you on a Kamailio SIP proxy server. Kamailio is the most popular and flexible sip server which can be used as a proxy, registrar and loadbalancer like Opensips. It works based on "kamailio.cfg" and it does not know anything than this file. This basic tasks seperate Kamailio from Asterisk Freeswitch because these act as "call process servers" instead of being loadbalancer or proxy.

Reverse Path Filtering on Linux (Redhat/Centos)

In this article, I will give an overview for use cases of reverse path filtering on linux environment. The situation that i have confronted with this feature was that running single kamailio sip server with two interfaces or vlans. Let's have a look;

Record-Route in SIP

In this blog, we’ll go through “record-route” header in SIP and also differences from “via” header that is used for responses. Shortly, the most important thing to defer this two header is that one of them is used for subsequent requests, the other one is used only for the response! Let’s have a detailed look:

WebRTC Overview

In this article, we will discuss to WebRTC which stands for Web Real-Time Communication that allows individuals and even enterprises to communicate via certain browsers DIRECTLY* without any third-party software and plug-ins. It’s not a solution, it’s a completely new technology. In other words, there is a world which lets that phones, TVs, computers can communicate on a same platform. It’s an open source and its codes is available for all at http://webrtc.org/. 

SIPP Performance Tool

Today, I’m goint to share my SIPP project that I have worked on it during last one month. Before begin this case, checking call performance on our voice network was my first goal. It can be said that being continuous of this project offers our team being awake against fail calls. Let me tell you requirements before start;

Port Mirroring for Recording SIP Calls by TekTape

In this article, I will let you know that how you record SIP calls on your network by TekTape tool which is produced by kaplansoft.  For some cases, requirements of your costumers can be met with a solution that you must get as outsource. I have encountered like this occasion just two days ago. Let me share my experience on this case;

Converting Cisco IP 7942G Protocol SIP from SCCP

As many people know, Cisco IP phones always use its own SCCP (Skinny Call Control Protocol) and SIP protocols with CUCM technology. In some different cases, engineers can be obliged to use other 3PCC servers in order to register their IP phones without CUCM. 

Traceroute

Traceroute is known as a network tool originally written by Van Jacobson to determine whether routing problems exist on the your network. It can be used to determine which path IP packets are taking to get from your computer to the remote computer.

Also, traceroute should not be used on a network without routers and it is not really useful unless there are at least two routers in the network. When you think international network (internet) has thousands routers and it make traceroute so useful for the internet. 

T.38 vs G.711

    Businesses migrating to VoIP find it desirable in order to move their fax traffic onto to the Fax over IP(FoIP). In this document, I’ll try to share with readers the performance two principal options for sending fax: T.38 fax relay and G711 fax pass-through. Except this methods, there are many fax settings such as fax speed, redundancy and error corection, can have significant impact on FoIP performance.

Voice Translation in Cisco GW

Voice translation rules have tripped up many an engineer, especially when troubleshooting an existing router that you may not have installed.  In those situations, it can be easy to misunderstand the subtleties of voice translation rules when quickly reviewing a production configuration.

There are some wildcard and their definition in below table;