Increasing VoIP services capacity

Using Kamailio & RTPproxy

This is a very advanced VoIP topic, but I'm sharing it here before I forget the hurdles which I faced while setting it up.

Introduction:
Setting up just one or two Media-Servers(Asterisk/FreeSWITCH/SEMS) isn't much of a headache but just one server is not enough once we start having more and more incoming call volume. Specially in larger deployments just one heavy duty server with multiple asterisks and freeSWITCH instances is never enough to handle everything efficiently and in fail-safe mode.

Current Media-Servers and their call capacity:
Remember a single instance of Asterisk on one regular server can handle 150~180 calls without any QoS issue. Running two or more instances on such a regular server may increase this capacity to around 300~380 concurrent calls but the fear of Single Point Of Failure is there all the time.

FreeSWITCH on the other hand claims to handle around 700~1200 concurrent calls on the same spec machines(rough figure to be on safe side, they claim more than this :P), but FreeSWITCH is a fairly complicated application to master for most of us.

SEMS claims to handle around 1800~2300 concurrent calls BUT..BUT SEMS is one of the hardest tool to configure and manage. Not to mention that it isn't promising any applications or services as Asterisk or FreeSWITCH are offering i.e Queues or Conferences, one may need to setup asterisk or FreeSWITCH behind SEMS. SEMS is an excellent choice for IVRs and announcements.

Problem at hand:
No matter how good you are in setting up Linux-HA, the biggest challenge of increasing VoIP media services capacity remains a big ? another big hurdle is assignment of Public IPs for each individual cluster of HA-Media Servers. Even if we manage somehow to handle capacity without thinking outside the Traditional-Asterisk-Solutions, reserving Public IPs still remains annoying problem.

Consider you've to buy Public IPs each time you need to increase VoIP services capacity along with headache of communicating with your services provider to add new IP into their call distribution list or provision traffic for new server.

Solution: 
So the solution lies in setting up a SIP-Proxy which acts as gateway distributor for all the incoming calls. This SIP-Proxy encapsulates a huge farm of Media-Servers. World only knows to communicate with you using just one Public IP belonging to the SIP-proxy.
All the Media-Servers are on Private LAN. Without any struggle anyone can put in 253 Media-Servers of all types in the Media-Servers zone. Simple and Easy ! isn't it.

RTP & SIP Handling on SBC

Using SIP-proxy as Load-distributor enables us to group all the media-servers in different resource types i.e 10 servers only handling IVRs, 13 serving Voicemails, 60 for Handling conferences only. 9 Servers as PSTN gateways etc.

SIP-proxy load-balancer will make sure each of these resources only get the requests for their service types only and all the load is distributed evenly(could be any other algorithm as well)

SIP-proxy will be sending Keep-Alive to all the Media-Servers and if any servers stops responding it marks it as Probing and continues sending KAs but not routing any calls to it until it becomes active again.

Whats NEXT:
In next post I'll try to show where to start from for setting up such an environment, what will be required, what problems I faced and their solution.

Special Thanks:
Following GURUs of VoIP from user's mailing list helped me alot in understanding and overcoming the problems I faced while setting this up.

Alex Balashov
davy van de moere
Klaus Darilion
Muhammad Danish Moosa
Muhammad Shahzad


Community list are definitely the best place to ask for minor help and understanding the problems.