Showing posts with label SRTP. Show all posts
Showing posts with label SRTP. Show all posts

Thursday, May 16, 2019

Switching to SIP? Four Facts to Check When Switching to SIP

Image result for SIP Trunk
SIP Trunking

SIP Communication in the business sector is getting traction as the businesses realize they can save a packet, sometimes up to 70% from traditional phone communications costs, by switching to SIP. So the SIP trunking is too gaining attraction to fill the technical needs of SIP communications.
SIP is traditionally a simple calling protocol yet there are some things to watch out when making the switch. Sangoma gives us Four most important and how to tackle them,

SIP Needs Security

Since SIP is in plain text, it doesn’t take an IT genius to decipher a SIP session (call). Without the proper security measures, anyone can run a packet capture tool (such as Wireshark) and extract audio from calls.

The good news is that this is an easy problem to solve with Secure Real-Time Transport protocol (SRTP) and a Session Border Controller (SBC), which is basically an application-layer firewall designed for SIP.

Make SIP (a) Priority

Between the media streaming and file sharing, chances are you have a lot of traffic on your network. If you implement SIP trunking on top of that without preparation, you may have delays in video buffering, slow email send speeds, or decreased audio quality on your VoIP calls.

To avoid this, make sure you utilize the standard QoS feature found in most business-grade routers and switches. The QoS feature will ensure your voice calls receive priority on the network, ensuring the available bandwidth is delivered straight to them first before less important network traffic.
Prior to SIP, Know Your Network

When switching to SIP, it’s important to have an accurate idea of how many concurrent calls your business makes. Review your call logs and understand the total number of calls your business makes, as well as how many concurrent calls are made at your busiest times.

SIP Bandwidth Needs 


The second thing to understand about your network is that SIP depends on bandwidth, and a lack of it can cause poor audio, dropped calls, and busy signals. Luckily, bandwidth is cheap and usually readily available!

Make sure you have enough to support the maximum number of concurrent calls your business requires. (To calculate the necessary bandwidth, simply multiply the maximum number of concurrent calls by Number of max concurrent calls x 100kb/sec = average bandwidth per call needed.) Simply add the bandwidth required to the amount you already use for business duties, and you should be good to go.

If You Fax Over SIP, Be Smart.

Faxing over IP can be very messy, particularly when it comes to the T.38 fax limitation. In very simple terms, a fax message cannot be compressed in the same manner as a voice packet. Packet loss with a phone call can result in poor audio quality, but you can often still understand what the person is saying. With faxing over IP, any packet loss can potentially cause the fax to fail completely.

That being said, if faxing is important to your business, then you should consider a reliable FoIP (fax over IP) solution such as FaxStation. With Sangoma’s FaxStation you can enjoy an analog fax experience without the added cost of an additional PSTN line.

You can find more information and solutions at Sangoma.

Sunday, August 09, 2009

FreeSwitch 1.0.4 Available For Immediate Download


FreeSwitch 1.0.4
I have been playing around with FreeSWITCH for a long while but recently I gave up playing and implemented the system for a small research team, that other open source VoIP Systems could not easily serve. As well as for the research team that is spread over a few countries and time zones and me personally the mod_skypiax gave the final push to put the system into production. We actually started working with the sources for this project much earlier.

The team at freeswitch.org has officially released the FreeSwitch 1.0.4. With this release FreeSWITCH brings many improvements in stability and security and some additions like the mod_skypiax.
Here are some of the functions and additions that I found interesting;

mod_opal
OPAL is one of my favorite projects and have been a follower for a while. Now FreeSWITCH works together with Project Opal to create mod_opal, which adds H.323 functionality to FreeSWITCH as well as IAX2 support.

mod_skypiax
My favorite, mod_skypiax brings capability to send and receive Skype calls from FreeSWITCH to Skype network. The Skype module (mod_skypiax) uses the native Skype client for the target operating system. Currently, Windows and Linux are supported. FreeSWITCH's ability to transcode and resample other high quality codecs allows for high quality Skype calls coming over SILK codec to connect to SIP endpoints, be they free SIP clients or Commercial SIP Phones.
This capability will certainly promote the capability FreeSWITCH's capability as a telephony switch.

ZRTP SRTP
ZRTP will allow to encrypt calls between endpoints or systems. The ZRTP uses SRTP to accomplish the purpose and is co developed with Philip Zimmermann of PGP fame, which I have been using for ages. I have not tested this feature yet but looking forward to do so soon.

mod_unimrcp
MRCP, or Media Resource Control Protocol, is provided with this module. This module utilizes the well-written UniMRCP library, authored by Arsen Chaloyen. The module is compliant with MRCP version 2 (SIP). MRCP allows for media servers to be separated from FreeSWITCH. Good module for tasks such as IVR or Text to Speech etc.

NAT Traversal Made Easy
NAT-traversal is important for SIP calls and is made easier with the automatic NAT handling feature included with the FreeSWITCH. Routers, switches and firewalls that support either UPnP or NAT-PMP can be polled by devices inside the network, allowing them to determine the external IP address. You can give up fighting with STUN now!
mod_nibblebill
FreeSWITCH now supports real-time pre-paid billing applications with mod_nibblebill. The name of the module is telling: it monitors calls in progress and "nibbles" away at the available credit of a pre-paid user's account.
In addition to above, the FreeSWITCH has following features;

  • Runs on Win32/MAC/UNIX
  • IVR Application API
  • 8kHz/16kHz/32kHz/48kHz Audio
  • Soft Conferencing
  • SIP B2UA/SRTP/TLS
  • SIP BLF/SLA/PBX Features
  • Presence
  • GoogleTalk
  • IPv4/IPv6
  • ENUM/ISN
  • Async Audio
  • Event/Logger Engine
  • Real Time
  • zRTP (libzrtp)
Free yourself with FreeSWITCH

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