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Quality of Service (QoS) FAQ's

What are the Quality of Service (QoS) features for the Allworx systems? [click here]

QoS features present in the Allworx product line (6x, 10x, 24x, 9102 and 9112)

- VLAN (with priority) support in phones.

- DiffServ tagging of all voice traffic: value 2E.

- Call Admission Control on a per-proxy and overall PBX basis.

- priority handling of voice vs. data on LAN-to-WAN passage.

If you are setting up QoS in a third party router the simplest way is to filter on DiffServ values.  All voip traffic from the PBX and phones uses a value of "2E", DSCP "46".

If you must use ports then UDP:5060 for SIP and range UDP: 16384 thru 32768 are used for RTP audio should get you by.  Other ports are used for boot time downloads and BLF message activity and shouldn't affect call quality.

Does the Allworx family of products support QoS? [click here]

Yes.

•  Priority passage of RTP audio from LAN-to-WAN

•  Ability to set call limits on a server-wide and per-proxy basis

•  Ability to specify jitter buffer settings on 9102/9112 handsets

•  Ability to specify codec for inbound/outbound calls on 9102/9112 handsets

•  Support of traffic segmentation using VLAN in 9102/9112 handsets.

•  Support of CoS within VLAN frame headers in 9102/9112 handsets.

•  Support of packet tagging for all VoIP traffic.

How does the Allworx family of products mark/tag voice related packets? [click here]

•  Equipment screening the DSCP field would recognize a value of “2E”. 

•  Equipment screening the ToS byte would recognize this as Priority value of “1”.

•  CoS bits within the 802.3 Q/P header (VLAN) can be set in the Allworx phones to 9102/9112 phones.

How much bandwidth is required per VoIP call? [click here]

90kbps or 35kbps of bi-directional bandwidth depending the codec being used on the call. 

- Refer to the Allworx VoIP White Paper for more details.

Which codecs are supported within the Allworx family of products? [click here]

G711 and G729a. 

- Refer to the Allworx VoIP White Paper for more details.

Why is call audio that is sent over the Internet sounds choppy at the other end? [click here]

Poor voice quality is typically a result of packet loss or jitter. 

•  Packets can be dropped or discarded at any point along its path of travel.  The most common point of discard is at the Cable/DSL modem or T1 router.  This is known as egress blocking. 

•  Jitter is the variable delay between packets; if packets are delayed too much they are not available when needed by the receiving handset for audio play out..