While the eventual goal is an end-to-end Next-Generation Network, it will be decades before legacy networks disappear. On the access side, this means that ongoing support for POTS telephone lines and DLCs may be a requirement; in the backbone network, interconnection with SS7 signaling and TDM trunks, 911 and operator services, databases for 1-800 and local number portability and CALEA, are all essential.
In addition migration will happen piecemeal in different carrier networks and individual service providers may support both next-generation and legacy networks in parallel. Therefore it is crucial that Next-Generation Network equipment provides support for legacy networks and that interworking between the networks is reliable and flawless. Service provides must also carefully plan migration strategies that include both introducing new services and support for legacy interfaces.
In addition migration will happen piecemeal in different carrier networks and individual service providers may support both next-generation and legacy networks in parallel. Therefore it is crucial that Next-Generation Network equipment provides support for legacy networks and that interworking between the networks is reliable and flawless. Service provides must also carefully plan migration strategies that include both introducing new services and support for legacy interfaces.
OSS Support
The existing PSTN has very extensive Operations Support Systems providing such functions as
• Flow-through provisioning
The existing PSTN has very extensive Operations Support Systems providing such functions as
• Flow-through provisioning
• Fault isolation
• Loop testing
• Alarms
• Performance monitoring
• Policy definition and enforcement
A Next-Generation VoIP network will need to offer similar levels of OSS support. In addition given the huge investment in existing OSS systems any new equipment will need to be integrated with these which may require support for protocols such as CORBA, SNMP, TL1, etc
VoIP networks also introduce new requirements such as the ability to dynamically measure end-to-end voice quality.
Bandwidth Utilization
In a VoIP network digitized voice is transported using real-time protocol (RTP). A typical voice sample is less than 100 bytes, but the combined headers are at least 40 bytes. For lower-bandwidth WAN links such as DSL or Cable, the header overhead is significant and reduces the number of voice channels or data bandwidth available. Given that one of the advantages of using VoIP is that it should be possible to use lower bit codecs to save bandwidth, a mechanism for reducing the overhead is required. The main approach to reducing the overhead is to implement compression for RTP, UDP and IP headers. However this requires a point-to-point link and the endpoints to maintain state for each compressed RTP flow.
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