Hi,
I'm implementing validation feature on RTP UDP audio packets.
I'm interested in the maximum values of the RTP packet per codec type.
Is there limitations size less than 1500 bytes?
The implementation covers the RTP resulting from sip/skinny/mgcp/h323 signaling.
I overviewed the rfc's 2327, 3551, 3555 and other online articles concerning
RTP. My conclusions are that there is no strict limitation, mostly there are
guidelines and recommendations.
There is a calculation for packet size:
BitRate (kbps) * packet size (ms) = packet size (bytes)
But if I put in it:
1. 64 kbps (for the g711 for example)
2. 200 ms (the limit of packet size according to QOS recommendations)
I get packet size = 1600 bytes. This is greater than 1500 bytes, so there is no
significant meaning for this limitation.
I would like to get some suggestions if in spite of this there is a way to
determine the maximum values of an RTP UDP packet size, per codec type.
The RTP codecs I'm interested in are (the defined in rfc 3551):
DVI4, G722, G723, G726, G728, G729, G729D, GSM, GSM-EFR, L8, L16, LPC, MPA,
PCMA, PCMU, QCELP, RED and VDVI.
Thanks
Zeev.K
_______________________________________________
Sip-implementors mailing list
[email protected]
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
