Re: [rtcweb] Consensus call regarding media security

Roman Shpount <> Thu, 29 March 2012 14:31 UTC

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Date: Thu, 29 Mar 2012 10:31:13 -0400
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From: Roman Shpount <>
To: Justin Uberti <>
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Subject: Re: [rtcweb] Consensus call regarding media security
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On Thu, Mar 29, 2012 at 10:00 AM, Justin Uberti <> wrote:

> This is FUD. Google+ Hangouts uses libsrtp for all of its calls, and over
> the billions of minutes of call time to date, we haven't seen any crash
> bugs that could be blamed on libsrtp. And we track this stuff pretty
> closely.
> This is not a FUD. Even with 100s of millions of secured minutes we are
pushing we see new libsrtp related problems on a weekly basis. I gave you a
reference to the Asterisk bug. This bug is addressed in sourceforge, but
present in the download library (1.4.4) that is included in a lot of the
products. This bug guarantees a crash in case RTCP and even small packet
loss are present.  You can try to run crypto_get_random in a loop until it
generates an error. Simple and easy to reproduce bug. There is probably
more, but it is outside of the scope of this list. If you want to do your
users a favor -- swap this lib out from your code. You probably have a much
better crypto utilities (random, AES) in other code you use, such as
OpenSSL. The rest is trivial to re-implement and it will make the result
product faster and more secure.
Roman Shpount