Polycom H340 - Manual (Page 12)

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Deploying SpectraLink e340, h340 and i640 Wireless Telephones BEST PRACTICES GUIDE
October 2010
Even with all of the known variables, there are many other vendor-specific characteristics associated with individual
APs that make it difficult to quantify the precise number of concurrent calls per AP, without thorough testing of
specific configurations. Polycom’s VIEW Configuration Guides identify the maximum number of calls per AP for
specific models that have been tested to be compatible with the SpectraLink handset.
With SVP, Polycom provides the ability to limit the number of calls per AP with a configurable setting in the SVP
Server. The “Calls per Access Point” setting limits the number of active calls on each AP and can be used to set
aside bandwidth for data traffic. Wireless Telephones in-call are free to associate with other APs within range that
have not reached the set maximum number of calls. Polycom requires this setting to be equal to or below the
maximum number of calls specified in VIEW Configuration Guides . It is still possible for the number of phones
associated to an AP to exceed the maximum number of calls as would be the case with any additional clients
associating to the same AP. The maximum number of calls per AP will simply control how many of those associated
phones will be able to enter into a call. Additional handsets beyond the maximum number specified will be forced by
the SVP Server to roam to a new AP that has not reached the maximum calls per AP. If no APs are available any
handsets beyond the maximum will display an error message indicating there is insufficient bandwidth to complete
the call.
2.3.2 Push-to-Talk Multicasting Considerations
SpectraLink i640 handsets provide push-to-talk (PTT) functionality using the Polycom-proprietary SpectraLink Radio
Protocol (SRP) ADPCM encoding. Because the PTT mode uses IP multicasting, all APs on the subnet will transmit a
PTT broadcast. This can be limited to only the APs that are handling one or more PTT-enabled handsets by
enabling the Internet Group Management Protocol (IGMP) on the wired infrastructure network. Ensure the LAN is
configured to propagate PTT (multicast) traffic to all WLAN APs used by the SpectraLink handsets.
When i640 handsets are deployed on a network with newer versions of SpectraLink handsets, some interoperability
considerations must be observed. The newer SpectraLink 8030 handsets have 24 PTT channels plus one priority
channel available. SpectraLink i640 handsets have eight PTT channels with no priority channel. When PTT is
activated on a network using a mix of handset versions, only the eight common channels will be available for the i640
handsets.
2.3.3 Telephone Usage
When the handset is used with traditional PBXs through a Telephony Gateway, the PBX interface will assemble
audio, packetize it, and release these packets at a preset interval. The PBX release interval is generally 20ms or
30ms. The SVP Server will receive these audio packets and release them to the network for delivery to the handset
every 30ms.
With a PBX release interval of 20ms, packets delivered to the handset by the SVP Server will have one audio
payload followed by a packet with two audio payloads. This pattern, one audio payload then two audio payloads, will
continue during the call. With a PBX release interval of 30ms, packets sent to the handset will have one audio
payload each. In rare occasions, a PBX may use a 40ms release interval. With this audio payload release interval,
packets delivered to the wireless telephone will have one large audio payload or no audio payload per packet sent to
the handset. The no audio payload packets and long time between audio (two SVP packets – 60ms) payload
aggravates any weakness (multi-path, retry packets, etc.) in the WLAN and will cause poor audio. Therefore,
whenever possible the PBX should be configured to use release intervals of 30ms or 20ms.
Because data rate and packet rates are constant with voice applications, wireless telephone calls may be modeled in
a manner very similar to circuit-switched calls. Telephone users (whether wired or wireless) generally tend to make
calls at random times and of random durations. Because of this, mathematical models can be applied to calculate the
probability of calls being blocked based on the number of call resources available.
Telephone usage is measured in units of Erlangs. One Erlang is equivalent to the traffic generated by a single
telephone call in continuous use. A typical office telephone user will generate 0.10 to 0.15 Erlangs of usage during
normal work hours, which equates to six to nine minutes on the telephone during an average one-hour period. Heavy
telephone users may generate 0.20 to 0.30 Erlangs, or an average of 12 to 18 minutes of phone usage in an hour.