[Live-devel] SRTP

RENOU Jean-Baptiste jean-baptiste.renou at thalesgroup.com
Thu Oct 31 03:32:45 PDT 2024


Classified as: {OPEN}

Hello,

I'm working with Jean-Claude, and we still encounter some bugs :

First, the update of ROC inside RtpSink  (calling RTPSink::setupForSRTP) is creating a new SRTPCryptographicContext and this will reset the ROC value to 0  (fHaveSentSRTPPackets is False, next SRTP packet enforces fSendingROC= 0). 
This breaks the authentication for both previous and new clients.

Additionally, when the SDP lines are renewed, I saw that the media description is different :

The first SDP is correct, for example :    m=video 0 RTP/SAVP 96
But, when renewing the SDP, the server RTP port is provided instead of 0, like :  m=video 26970 RTP/SAVP 96
Then, the client is using that value as client RTP port instead of looking for a random free port (method MediaSubsession::initiate) if not compiled with "IGNORE_UNICAST_SDP_PORTS". 
This may be a problem as there's no guarantee that the port value is available on client side.

Jean-Baptiste Renou


{OPEN}

-----Message d'origine-----
De : live-devel <live-devel-bounces at us.live555.com> De la part de Ross Finlayson
Envoyé : mercredi 30 octobre 2024 17:47
À : LIVE555 Streaming Media - development & use <live-devel at us.live555.com>
Objet : Re: [Live-devel] SRTP

> On Oct 30, 2024, at 10:55 PM, SAGET Jean-Claude via live-devel <live-devel at us.live555.com> wrote:
> 
>  The ROC is now well updated but in method RTPSink::setupForSRTP(Boolean useEncryption, u_uint32_t roc) the SRTPCryptographicContext is deleted and still referenced in RTCPInstance and used in RTCPInstance::processIncomingReport and RTCPInstance::sendBuiltPacket.

Jean-Claude,

Many thanks for your patience.

I have just installed a new version (2024.10.30) of the code that should fix this.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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