Mixer In RTP
It may so happen that all participant in the conference do not have connection of same bandwidth. Now the question is how do they participate in the conference.
One solution could be that all of them use sane lower bandwidth. But this leads to reduced-quality audio encoding.
A smarter solution exists in the use of a RTP level relay called a mixer. A mixer can be placed in a lower bandwidth area. This mixer resynchronise incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, Mixes the reconstructed audio stream to a single stream, translate the audio encoding to a lower bandwidth one and forwards the lower bandwidth packet stream across the low-speed link. The following gives a Graphical representation.
The mixer puts its own identification as the source of the packet (SSRC) and puts contributing sources in the CSRC fields.
Mixers have other applications too. for e.g. a video mixer that scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene.
Translators in RTP
A problem occurs if one or more participants of a conference are behind the firewall which won't allow an ip packet containing an RTP message to pass. For this situation, translators are used
Two translators are installed, one on either side of the firewall with the outside one funnelling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site's internal network. The illustration is given below.
The translator do not change the SSRC or CSRC unlike the mixers.
References:
http://www.siptutorial.net/RTP/mixer.html
No comments:
Post a Comment