Subject: FW: Need for simple RTP multiplexing for trunking gateways

From: "Rajeev Gupta" <[email-masked]>
Date: Wed, 3 Jan 2001 19:32:47 -0800

-----Original Message-----
From: Rajeev Gupta [mailto:[email-masked]]
Sent: Thursday, December 21, 2000 4:22 PM
To: [email-masked]
Cc: [email-masked]
Subject: Need for simple RTP multiplexing for trunking gateways

Consider a VoIP network with a set of trunking gateways connecting the PSTN
over a VoIP core network. Each gateway has thousands of PSTN ports and uses
G.711 to get high quality voice over IP. Between each pair of gateways,
there are hundreds or thousands of calls being trunked. The core IP network
does not today provide MPLS capability. In such a scenario, it is required
to provide a high quality, low overhead solution for VoIP.

The current RFCs and I-Ds suggest to use RTP/UDP/IP between the gateways. To
reduce header overhead, it is suggested to run TCRTP. But that has the
following disadvantages:

- it requires creating tunnels using PPP/L2TP
- it recommends using PPP multiplexing to reduce tunnel overhead
- it requires running a compression/decompression state machine
- Further complexity is added due to loss recovery on high delay, high loss
WAN links

The above problems are compounded when you have thousands of calls running
between gateways.

A much simpler mechanism would be to allow simple RTP multiplexing using
standard RTP headers. This would have the advantage that:

- it would use standard IP/UDP/RTP headers, requiring no tunnelling or
special compression state machines

- it would use a small channel id to multiplex hundreds of channels into the
same packet. Trunking gateways can multiplex hundreds of simultaneous
samples into the same IP packet, reducing overhead to a negligible amount

- channel ids can be signalled very easily as just another media parameter
in the SDP when setting up a new call

- with silence suppression turned on, this would yield bandwidth
efficiencies greater than TDM with voice quality (G.711) better than
compressed vocoders (G.72x)

- since bandwidth efficiencies are high, we can even reduce the
packetization period to less than the default 10ms, to get even higher voice
quality (due to reduced latencies and reduced loss rates, similar to
"oversampling" effects)

I am currently working on a draft to propose such a solution and would be
interested in hearing from others on their perspectives.

Note this is not related to Voice over MPLS or similar proposed work, in
that this is designed specifically to work over standard IP, which may or
may not run over MPLS.

Thanks.

Rajeev Gupta
Senior Director, Technology & Standards
empowerTel Networks
475 Sycamore Drive, Milpitas, CA 95035
Phone: +1 408 519 7136
Fax: +1 408 519 7420
mailto:[email-masked]
http://www.empowertel.com