Review – SIP softclient CSipSimple

When I teach a SIP Essentials class, I always get asked about SIP clients. So many are available, so I’d like to begin asking the question, which ones are stand out? Which ones should we all flee from?

In this posting I’ll be reviewing CSipSimiple. CSipSimple was tested on my Nexus7 (Jellybean) on Oct. 31, 2012. The app is not available in the iTunes marketplace.

In a few sentences; CSipSimple is a free app, with a clean layout, no ads, many configuration options, as well as a handful of useful call features. I found it rather straightforward to configure a user account and register to one of my Asterisk boxes; I was making my first test call within a minute. If you’re looking for a SIP client to keep on your Android device, don’t hesitate to reach for CSipSimple.

The most basic registration option requires you only enter the most basic options to register CSipSimple: An account name, User, Server (IP) and Password associated with the SIP account.

Features I found packed with this free SIP app:

  • No advertisements

  • Quick setup and easy registration of a SIP User

  • Built in ‘Account Wizards’ make it easy to connect to a number of VoIP providers if you don’t run your own SIP box

  • Supports ICE & STUN

  • Supports TLS, SRTP & ZRTP

  • Customize phone theme, ringtone & answer options

  • Call transfer (no call forwarding)

  • Supports multiple calls

  • Record calls & review call log

  • Echo cancelation (I was unimpressed)

  • Contact integration

  • Compact SIP

  • Basic codec support (no video support at this time)

Supported codecs are as follows:

PCMU / PCMA (aka g711u/a); speex; g722; gsm; iSAC; SILK; G729; AMR (depending on device) and as extra plugin : OPUS; g726; g722.1; codec2

A few screenshots of CSipSimple running on my Nexus7.

(Above) Screenshot of CSipSimple on my Nexus7 - registration screen

(Above) Screenshot of CSipSimple on my Nexus7 - dialing pad

(Above) Screenshot of CSipSimple on my Nexus7 - an active call.

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A basic Asterisk v1.8 sip.conf configuration