[SIPForum-discussion] Voice Quality in SIP

[McDowell, John T]

Voice quality references the bearer traffic, while SIP and BICC both reference control.  The quality of your bearer traffic will depend on many factors, none of which are dependent on your control protocol.

If migrating from traditional telephony (as would likely be the case if you're using BICC) to IP telephony, there are many thing to consider.  Traditional telephony reserves bandwidth for each call, while IP telephony typically does not.  It is the reservation-less nature that provides both the benefit and hazard of IP telephony.  It provides the ability to dynamically route/allocate resources, but also brings up the possibility that bearer traffic could be handled inappropriately, or not at all.

You will want to consider how your bearer traffic will be handled.  Meaning, what codecs will be used, will QOS be engaged, what is the level of latency you can tolerate, will your partners have terms of service they must follow, will you need to specify the metric types (MOS Scores), etc.

John McDowell
Network Services
Highmark Inc
(w) 717-302-5995
(c) 717-319-6695
(h) 717-712-0971

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of SARKER, ROHAN (ROHAN)
Sent: Wednesday, January 13, 2010 5:12 AM
To: discussion at sipforum.org
Cc: rohansarker1 at gmail.com
Subject: [SIPForum-discussion] Voice Quality in SIP

I have one more question in this regard.

Is voice quality in SIP better than in case of ISUP or BICC??

Second question: How do we do Acceptance Testing for SIP Trunking? Is there any document or Test Plan in this regard?











Thanks and regards

Rohan Sarker

Deputy Manager

Alcatel-Lucent

DAKC, Navi Mumbai

M:+919320496780

R:+919903051236
O:+912230372803


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