QoS DSCP for Call-Signaling

Call-Signaling Traffic

The following are key QoS requirements and recommendations for Call-Signaling traffic:

Call-Signaling traffic should be marked as DSCP CS3 per the QoS Baseline (during migration, it may also be marked the legacy value of DSCP AF31).

150 bps (plus Layer 2 overhead) per phone of guaranteed bandwidth is required for voice control traffic; more may be required, depending on the call signaling protocol(s) in use.

Call-Signaling traffic was originally marked by Cisco IP Telephony equipment to DSCP AF31. However, the Assured Forwarding classes, as defined in RFC 2597, were intended for flows that could be subject to markdown and - subsequently - the aggressive dropping of marked-down values. Marking down and aggressively dropping Call-Signaling could result in noticeable delay-to-dial-tone (DDT) and lengthy call setup times, both of which generally translate to poor user experiences.

The QoS Baseline changed the marking recommendation for Call-Signaling traffic to DSCP CS3 because Class Selector code points, as defined in RFC 2474, were not subject to markdown/aggressive dropping. Some Cisco IP Telephony products have already begun transitioning to DSCP CS3 for Call-Signaling marking. In this interim period, both code-points (CS3 and AF31) should be reserved for Call-Signaling marking until the transition is complete.

Many Cisco IP phones use Skinny Call-Control Protocol (SCCP) for call signaling. SCCP is a relatively lightweight protocol that requires only a minimal amount of bandwidth protection. However, newer versions of CallManager and SCCP have improved functionality requiring new message sets yielding a higher bandwidth consumption. Cisco signaling bandwidth design recommendations have been adjusted to match. The IPT SRND's Network Infrastructure chapter contains the relevant details, available at:http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guides_list.html.

Other call signaling protocols include (but are not limited to) H.323, H.225, Session Initiated Protocol (SIP) and Media Gateway Control Protocol (MGCP). Each call signaling protocol has unique TCP/UDP ports and traffic patterns that should be taken into account when provisioning QoS policies for them.
Post a Comment

Popular posts from this blog

Pairwise Master Key (PMK) vs Parewise Transient Key(PTK) vs PseudoRandom Function(PRF) vs GTK (Groupwise Transient Key)

DSSS(直接序列展頻技術) vs OFDM(正交頻率多重分割)