[FFmpeg-devel] RTCP question

Michael Niedermayer michaelni at gmx.at
Wed May 14 02:55:14 CEST 2014

On Thu, Apr 24, 2014 at 05:05:17PM -0600, Fred Rothganger wrote:
> I am dealing with a use-case that may involve long pauses in
> streaming. Because of this, it may be useful to ensure that a fresh
> RTCP SR packet is sent when streaming restarts.
> Here is a code fragment from rtpenc.c, round line 500:
>     rtcp_bytes = ((s->octet_count - s->last_octet_count) *
>         RTCP_TX_RATIO_DEN;
>     if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
>                             (ff_ntp_time() - s->last_rtcp_ntp_time >
> 5000000))) &&
>         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
>         rtcp_send_sr(s1, ff_ntp_time(), 0);
>         s->last_octet_count = s->octet_count;
>         s->first_packet = 0;
>     }
> Notice the AND between the byte-ratio and time-elapsed conditions.
> Would it make more sense to make this an OR? IE: shouldn't we send a
> fresh SR message if *either* condition were true, as opposed to both
> simultaneously? As written, there could be a pause for 10 minutes,
> but if not enough bytes have been sent yet, it may be several
> packets before a fresh SR message is sent.
> Am I missing something here? I suppose the intention is that if the
> overall bandwidth use is very low, then don't fill it up with SR
> messages. I would argue that in that case, you won't add much to
> bandwidth usage anyway.

what does the RFC say about the RTCP SR interval ?
if theres a max time specified then the code should honor that
independant of how many bytes have been transmitted

Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

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