[SIPForum-discussion] ITU recommendation of 150 ms Round Trip Latency

sreekant nair sreekant_nair at yahoo.com
Wed Jul 25 12:25:42 UTC 2007


This is regarding the ITU recommendation that the one way latency of any signal/voice packet for a call should not be greater than 150ms. 

In the system being developed, the voice codec has a packetization interval of 30 ms and so we fall within this recommendation while transmitting RTP packets. However during testing we noticed that in the reverse direction (incoming to our system), we were receiving RTP packets with a 60ms packetization interval. Basically the remote server was clubbing two packets and sending them together since the bandwidth of the link was high. 

The question I had was is it possible to extend the packetization interval to say 90ms, and transmit the RTP voice packets over the IP Backbone. Since the latency of the IP network is non-deterministic, what would happen if the delay was such that it exceeded the ITU recommendatation of 150 ms. Since we have already waited for 90ms to generate the packet, we are left with 60 ms for the packet to reach the destination and in all likelihood, 60ms would not be enough. 

In your experience, what would an acceptable value for the delta be so as to not compromise too much on the voice quality. 

Appreciate any thoughts in this regard. Thanks in advance. 

Sreekant Nair

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