Session Initiation Protocol 101
Last week I was at the Aspect Consumer Experience conference and attended a session on SIP - here are some notes from the session providing you with all the basic things you need to know about SIP
Steps in VOIP are similar to a phone call
· Caller specifies
· Called Party receives
· Communication happens
Importance of SIP
Ø New hot standard for VOIP
Ø Almost every vendor moving to it
Ø Focus on P2P connections without controller authentication
Ø Not only for voice but also IM, audio, video, data -> this is a key differentiator
Ø Supports addressing by URI -> phone, name, email
Ø Supports presence for user and not just for the device
Ø Vendors can add proprietary extensions as part of VOIP call – device can ignore data if it is not understood
Benefits of VOIP
· Runs over the same wiring as the computer network
· Free – no toll charges for local, international, long distance
· Mobility and Instant communication
· Not just voice – multimedia enabled
· Presence enabled
· Toll bypass – connect to agents anywhere
· Security – run calls on private network, VPN, encrypted network
SIP components -> Session Border Controller (SIP Firewall), Media Gateway, Media Server, User Agent (End point), SIP proxy (routing and additional resolution). Registrar allows for UA to register with SIP proxy and enable plug and play capabilities. Redirect Server aids redirecting and is used by the proxy. The Presence server is used to determine the state of UA. SIP proxy can be of 2 types – Pass through, B2B UA (provides additional capabilities – call forwarding, conferencing)
Each Request type has an appropriate response associated, for e.g. for an invite the response might be OK or Trying or Ringing.
· SIP follows the RFC 3261 specification
· Codec – coder/decoder -> takes audio and encodes for sharing over the internet,
o most common is G711 which is equivalent to PSTN quality (64 kbps -> 64 kbps)
o G729 -> compresses (64 kbps -> 8 kbps)
o G722 -> better quality than PSTN, hi-fi voice quality
· Voice packetization – broken-down, sent and then recompiled
· Network considerations
o Start with big pipes (minimum of 100 Mb on switched LAN)
o VLAN – segregation of traffic (ensures bandwidth for audio)
o Firewall with network (may cause a delay, open application ports, could pose additional translation issues)
o Distance affects audio delay
· Bandwidth requirements
o Bandwidth calculation tools available
o Compression codec
o Silence suppression (can cause clipping)
o Header compression
· Audio quality
o Jitter buffers – keep a set of packets to allow for smooth playback, recorder packets, deal with packets coming in at faster or slower rate
o Dynamic jitter buffers – changes buffer based on how packets are received
o Echo cancellation
o Delay of less than 150 ms is considered good
o Clipping of greater than 20 ms is perceptible to humans
· Presence
o SIP is provided via SIP Instant Messaging Presence Leveraging Extensions aka SIMPLE
o Can provide rules to govern presence
o Intl chars supported
o IPV6 supported
· Authentication, Security and Encryption
o Phone/users identify themselves
o VPN for remote users
o Encryption not widely supported as will cause delay
· High availability with SIP – points of failure are as follows
o PSTN communication, SIP proxy, IP phone, IP PBX, Network connectivity, Power
· SIP trunking – remove media gateway, directly from carrier, reduce jitter, pass non-telephony data, IP multimedia sub-systems – send application info
· Cons of SIP
o As good as the network
o Emergency service is an issue
Other applications running on same network will cause lag
Comments
- Anonymous
May 19, 2007
Quite interesting informations. But today, Microsoft delivers Windows Mobile 6 including the VOIP and the SIP protocol managment. Where could we get more precise informations on transaction or settings to use the WM6 Voip properly? I wen through the msdn, but nothing helping about the sound configuration of the Voip, about the voip status managing, among other problems Thanks.