Audio synchronization between individual RTP audio streams

Vitaliy Lazarev vlazarev at
Tue Nov 9 16:08:47 UTC 2021

Hi Tim,

Thank you for the response.
We found some relevant RFCs that may help with our problem, as you
mentioned. I guess we're talking about the RFC 6051 and the
so-called in-band synchronization mechanism.
We found the issue that was opened about 4 years ago:
I guess it wasn't merged to any branch or version of Gstreamer yet. Is this
the correct issue thread and patch? Are we on the right way?

Thank you,

*Vitaliy Lazarev*
Software Engineer,
notAnotherOne <>

E-mail: vlazarev at
T: +7 911 296-73-93

On Fri, Nov 5, 2021 at 4:33 PM Tim Müller via gstreamer-devel <
gstreamer-devel at> wrote:

> Hi Vitaliy,
> > We have a system with one RX pipeline that is running on RPi4 and
> > listening to several UDP sources with different ports, and near 10
> > individual TXs. All of the devices are located in the same room. TXs
> > are sending their own RTP stream over Wi-Fi to RX simultaneously +
> > RTCP with SR and SDES. And after some time due to network connection
> > issues and other factors, these TX audio streams are running out of
> > audio sync (from 100ms to 1s audio lag).
> > We found a similar issue, and it says that if there is no RTCP
> > connection between RX and TXs, there will be no synchronization. But
> > this issue is about hosts that are not located in the same physical
> > space.
> > Is RTCP usage can help to prevent audio lag between individual RTP
> > streams (maybe feedback with RRs)? Can it be solved only using RX
> > side with some manipulations with pipeline elements (rtpbin NTP/RTCP
> > sync options), RTP timestamps, or something?
> > Is this even possible to get a several milliseconds synchronization
> > between many RTP audio streams, that are located physically in one
> > room?
> The general "problem" with RTP is that the timestamps in the RTP
> packets are offset randomly without any absolute reference or base. A
> receiver will typically just record the timestamp on the first packet
> and map that to some local 0 base time and then interpolate from that.
> If a sender sends audio and video or multiple audio streams, that might
> mean that the receiving streams could initially be out of sync.
> Sender report (SR) RTCP packets provide extra information from the
> sender to receivers. They contain mappings between RTP timestamps to an
> "ntp time". This then provides a receiver with a common time base for
> all streams coming from a sender, so it can use that to offset
> audio/video streams accordingly and achieve lipsync. A receiver doesn't
> necessarily know what these ntp timestamps refer to though, so it can
> only use it to sync the incoming streams relatively, but not to map it
> to an absolute or local time or clock.
> Now if you have multiple senders there's another problem: Different
> machines will be using different clocks, and clocks drift over time.
> So what you want to do is ideally make all devices (or at least all
> senders) use a common clock. This can be an NTP clock or a PTP clock or
> a GstNetClock tracking a local GstNetTimeProvider.
> Once you've done that you make the senders use that clock for sender
> report ntp time (for bonus points configure sender to use capture time
> instead of send time for SRs, but depends a bit on your hardware and
> drivers how well that will work).
> Then all senders will basically be using the same clock/time as
> reference for their ntp time stamps, and the receiver can correlate the
> streams from the different senders.
> There are also RFCs for RTP header extensions that allow senders to put
> ntp timestamps into each packet they send out which allows for rapid
> synchronisation (no need to wait for a Sender Report), but you still
> need for senders to agree on a clock/time reference in order fo this to
> be useful.
> I believe if you configure the AVPF rtp profile it will send a SR
> immediately at the beginning instead of only sending it after a few
> seconds.
> Good luck!
> Cheers
>  Tim


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