There is no denying the fact that custom development in software gets you price and performance benefits. Take VoIP. This segment is maturing and has several open source platforms that you can choose for tailored high performance applications. You can pick Freeswitch, Asterisk, Kamailio, OpenSIPs or WebRTC to get benefits of each one in a compact package. You could have VoIP apps like IP PBX, conferencing solutions, contact center software or SMS broadcasting built on any one or combination of these platforms.


Asterisk is quite mature and its current iteration supports a variety of codecs and protocols. Being open source, it is free and you pay only for the custom VoIP solution development service. It is flexible and easily lends itself to custom development of VoIP apps like IVR, IP PBX and conferencing solutions. You can get developers to build voicemail server, VoIP gateway or IVR server based on Asterisk. While IP PBX is a common application that works superbly on Asterisk, custom development will get you purposed apps for phone verification, polling, billing and click2call. Incorporating Click2Call on your website, for instance, considerably improves conversion, especially when websites are used on mobile platforms. Conferencing is another area where you can have customized and simplified options instead of subscribing for features you do not need. The result is compact, fast apps with several layers of encrypted security.


Asterisk is used in over 150 countries but there is no reason to ignore Freeswitch. It grew out of Asterisk and is better in some areas, such as concurrent handling of higher number of calls. One advantage of Freeswitch is that it can work on any Operating System and lends itself admirably to building anything from a simple IPPBX to SIP trunks and session border controller. VoIP developers will prove immensely useful when you want to interface legacy VoIP hardware to software or to extend system functionalities through applications written in Java, .NET, C, Python or Javascript. If codec and protocol interoperability is a prime consideration then Freeswitch is an excellent choice. Likewise, if reducing bandwidth is important then Freeswitch proves to be better. Your VoIP developer should recommend Freeswitch or Asterisk based on what you wish to achieve.


Google gave the world WebRTC in 2011. Unfortunately it is not like an app you can use directly though it integrates with the browser for real time audio-video chat and media exchange. You do need VoIP solution developer to come up with API integrations and IETC protocol implementation for workable solutions. In the current physical distancing and remote work environment you will WebRTC a perfect fit for secure conferencing. The Opus audio codec allows high quality audio and VP8 video codec ensure jitter-free video stream. With WebRTC you can bypass traditional VoIP tech and get going at a very low cost. Since it works on a browser and can handle multiple endpoints it is ideal for audio-video communications. Startups should take a closer look at getting WebRTC app for their business instead of going the IPPBX route. You get price and performance benefits. However, even those who are using VoIP based IPPBX can easily integrate WebRTC to enhance use experience in areas such as call centers, education and business.


Talk of SIP servers and the name Kamailio pops up. It has an acknowledged leadership position in this segment. Kamailio SIP servers have all the features of OpenSER and SIP Express Router with some more additions. Carriers and VoIP services that need custom SIP servers will find that engaging VoIP solution providers specializing in Kamailio will give them rich dividends in a powerful, compact but modular solution for database interface, authentication, load balancing, NAT traversal and flexible architecture implemented to suit your specific operations model. Kamailio does offer plenty of features and that in itself points towards custom Kamailio solutions to get price and performance gains. Some ITSPs tend to migrate to Freeswitch or Asterisk when they find it difficult to use Kamailio based SIP servers. In such cases, calling in experts in Kamailio can resolve the situation and avoid the expensive migration process.


Kamailio is inarguably high end for ITSPs and carriers but you also have the option of OpenSIPS proxy server that will work well even in demanding environments with the ability to handle millions of concurrent calls. There is no lack of features. You have SIP router, registrar, application server, session border controller, front end, NAT traversal and other features. It may be open source and free which also gives choices of over 120 modules for application building. Startup VoIP service providers and existing telecom carriers might find it worthwhile to consider custom VoIP solutions based on OpenSIPS. It could provide the best price-performance mix.

Custom VoIP development is not just about branding or features. You get more by way of reduced load on infrastructure and higher performance at a lower cost.

Author Bio:

Hiten Dudhatra is a Team Lead - Digital Marketing at Ecosmob Technologies Pvt. Ltd. He likes to share his opinions on IT & Telecommunication industries via guest posts. His main interest to write the content for Asterisk, FreeSWITCH, OpenSIPS, Kamailio & WebRTC. @hitendudhatra

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