Microsoft TechEd – June 9th, 2010

Yes; New Orleans is still hot.  When I leave the building my glasses fog up and my shirt gets drenched within minutes.

From a blogging perspective, today I switched to using my work laptop and Microsoft Live Writer – I need some formatting in these blogs.  I love my iPad but it just isnt ready for formatting prime time in blogging.

Here are my notes from a session I took yesterday afternoon:

Microsoft Communications Server 14: Voice Architecture and planning for high availability.

What constitutes good voice quality?
Starting point for most people is the PBX phone "narrow band audio".  Good voice quality is highly personal and context sensitive.   Up to a point, users will accept lower voice quality given other advantages:
      -cell phones
      -internet voip

Understanding the challenges
   Call reliability
      Dropped calls
      Failed calls
   Audio quality
      Broken up audio
      Delayed audio
      Distorted audio
      Low volume
      Noise
      Echo
      One way audio
4 big areas of concern
   1.) Network
   2.) Core performance (application)
   3.) Gateways
   4.) Devices

This is a great goal setting for network performance:

Network performance goals

   1.) Jitter average < 10 ms

   2.) Jitter max < 80 ms

   3.) Packet loss < 10%

   4.) Network latency RTT < 200 ms

Anatomy of a UC audio call
   SIP
   SRTP / RTCP
   FEC –> Forward Error Correction
      Allows for audio healing for missing packets (sounds metallic)

Audio/video bandwidth usage
   1.) How much bandwidth is required is determined by:
         a.) Codec choice
         b.) Network performance
         c.) Poor network performance results in redundant encoding of audio
         d.) Voice activity and video content
         e.) Media endpoints actively manage distribution of bandwidth across UC modalities
     

   Office communicator prioritizes audio first and distributes the remaining bandwidth to app sharing, video, and file transfer.

Codec choice     

   1.) Chooses the best quality codec and video resolution for the available bandwidth 
   2.) May dynamically change codec choices during a session
Audio/video bandwidth profiles
 

Codec typical bandwidth
  Rtaudio 8 kHz - 25.9 kbps
  Rtaudio 16 kHz - 34.8 Kbps
  Siren - 22 kbps
  G.711 - 59.8 kbps
  G.722 - 42.8 kbps
  Rtvideo - CIF 15fps - 203 kbps
  Rtvideo - VGA .....

Audio/video bandwidth controls:
   - End user maximum allowed bandwidth per modality 
   - Applied whether or not bandwidth is available
   - Configured via in band provisioning at signon
   - Wide area network link bandwidth policies (call admission control) CAC
    - Applied dynamically when session crosses network
Call admission control

   New policy server role introduced in wave 14
      Admins create logical network sites
      Enforce policies between sites
         Bandwidth available for audio/video   
         Max allowed per session
         Rerouting behavior when exceeded
      Seamless support for roaming users
      Allows internet to be used for overflow of traffic
          Avoids PSTN charges
          Support failover for video

What about application sharing?
   Bandwidth used by application sharing is highly dependent on session content and screen resolution
   Traffic is bursty - zero in steady state then bamn, spike
   Tcp based sessions
   End user policy limits available to cap spikes

Network QoS - DiffServ
   Where do we recommend Quality of Service
      When right provisioning is not possible and so constrained WAN links
      Audio prioritization already deployed for other VOIP solution
   Differentiated services code point (DSCP) - field in an IP packet to assign levels of service for network traffic.
Environmental factors
   Windows 7 only environments - can use windows policy based QoS
   OC communicator phones mark at endpoints.

VLAN discovery is changing in 14 - using LLDP plus e911 and power Mgmt

VPN's - use a split tunnel approach and IPSEC – just don't use; causes delays, setup failures, mid call drops

Session resiliency and recovery
  Signaling plane - tcp
  Media plane - udp or tcp
   Some middle box like hlb's causes tcp resets.  Need to keep them in sync

Core performance, devices, and gateways
   Media by-pass - calls can go directly to a PSTN gateway
      -Improves audio quality and reduces number of servers
   Default codec changing from siren to g.722 for quality in wave 14
  

Wave 14 brings with it a new suite of phones including conference room based phones

New feature - window reporting back to the user that their device may be causing choppy audio. - or it will say "network connectivity i causing audio quality issues"     

Notes taken on my iPad, published with Live Writer.