Posts Tagged 'SRTP'

PJSIP version 2.5 is released with support for Opus and WebRTC AEC

PJSIP version 2.5 is released with main focus on Opus codec and WebRTC AEC integrations. The PJSIP bundled libsrtp package has also been upgraded to version 1.5.4 which brings a higher level of media security via AES-256 crypto suites.

As usual the release also includes several enhancements and bug fixes, please see the Release Notes page for more info and grab the source code from the Download page.

PJSIP 0.9 is Released: Audio Latency, TURN implementation, IPv6, G.722, and More

Finally, after months of delay, PJSIP version 0.9.0 is released. This has been the longest gap (8 months) between releases, and consequently it has the most modifications in it (there have been 100+ tickets done on this release).

Some of the new features in this release:

  • many improvements in the audio, to reduce audio latency, to have better compatibility with more target platforms (Windows Vista issues have been fixed, as well as sporadic error reports for ALSA), and to maintain the audio quality against impairments such as clock drifts, bursty sound device, and of course, packet loss. Compared to version 0.8, I think we’ve improved audio latency by few hundred milliseconds.
  • support for TURN-07 in PJNATH, either as standalone client/server library, standalone client/server application (for testing purposes), or integrated with ICE-19. Just as we were the first to release open source ICE library, I think this is also the first open source TURN implementation out there. Unfortunately we haven’t had time to update it to TURN-08 as this draft was released late during our QA phase, but we’ll update it as soon as possible.
  • fixed the ICE offer/answer rules.
  • support for IPv6.
  • support for Secure RTP (SRTP)
  • better support for Windows Mobile target. We have new and more usable sample application (PocketPJ) and GSM and Speex codec should now be available for this target.
  • better support for Symbian S60 target. There is a more thorough Symbian tutorial available, and GSM and Speex codec should now be available for this target too.
  • implementation of G.722 codec.
  • support for RTCP Extended Report (XR)
  • and many more.

For more information, start from PJSIP download page. Get it while it’s hot!

Securing VoIP: SRTP Support in PJSIP

PJSIP now has SRTP support in SVN trunk (hurray!). For more information about compatibility, how to use, and what have been done, please see the SRTP wiki in
http://trac.pjsip.org/repos/wiki/SRTP

We’ve tested it against couple of phones that support SRTP and it looks good (apart from SRTCP, which one phone doesn’t support and the other we don’t think it implements SRTCP correctly). If you have phones which support SRTP, it would be great if you could give it a try, I’ll be thrilled to hear your result.

Enjoy it while it’s hot!


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