[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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