[SIPForum-discussion] ITU recommendation of 150 ms Round TripLatency

Agonias, Richard L. (Digitel-GSM) Richard.Agonias at digitel.ph
Thu Jul 26 01:10:56 UTC 2007



In relevance to Sreekant's query, I want to add the following:

1.	Is the packetization interval of 30ms corresponds to the AMR12.2
frame which consists in 244 bits? If this is the case,
2.	AMR12.2 has a built in VAD/SID functionalities, how will this
affect the roundtrip latency? 


Thanks and best regards,

Richard Agonias






From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of sreekant nair
Sent: Wednesday, July 25, 2007 8:26 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] ITU recommendation of 150 ms Round



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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