Translators and Mixers In RTP Sessions

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Translators and MixersIn RTP Sessions

• By
• Michel Shahan

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VoIP

• Need for a protocol to send real-time media.
• Interactive. (voice calls)
• On-demand. (video clips)

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RTP and RTCP

• Real-Time Transport Protocol (RTP)
• On top of UDP
• Transferring audio and video
• No designated port. (even)
• Real-Time Control Protocol (RTCP)
• Reports information about the session.
• No designated port. (odd)

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Translators and mixers

• Intermediate RTP level systems
• Placed anywhere in the network. Usually near the
  end-systems to be more efficient.
• Generally not the same physical device
• Necessary in some RTP session
• Conferencing
• Traverse Firewalls

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Translators and mixers

• Translators (invisible)
• Traversing firewalls (multi/uni- cast)
• No designated port number
• Re-encode
• Lower/Higher quality/bandwidth use
• CODEC (COder DECoder)
• Bridge different transport protocols
• UDP?? ATM
• Mixers (visible)
• Mix several streams into one single stream
• Re-encode

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Understanding the RTP

• Understanding the Header
• Developed with Translators and mixers in mind

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RTP header fields

• CSRC Count, 4 bits
• The number of CSRC fields. Maximum 15
• Payload Type, 7 bits
• Identifies the default payload type.
• RFC 3551
• Sequence Number, 16 bits
• The order the packets are sent.

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RTP header fields

• Timestamp, 32 bits
• Denotes sampling instant for play out scheduling
  (random)
• Synchronization Source (SSRC), 32 bits
• Identity for sources in the session. (unique)
• Grouped by receiver for playback
• Contributing Sources (CSRC), 32 bits
• Mixed streams SSRC.
• Identify talker

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Translators receiving RTP packets

• Changing Payload type
• Decode payload, encode payload, Change payload
  field
• Lost packets from source to Translator
• Change sequence number
• Combining packets from same stream
• Less overhead
• Change the Sequence number
• Invisible

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Mixers receiving RTP packets

• Combining different streams
• Possibly different encoded streams.
• buffering
• New timestamp
• New unique SSRC
• CSRC Count changed
• Adding CSRCs

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Combining translators and mixers
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Understanding the RTCP

• Provides the RTP with congestion control
• Sent in compound Packets
• 5 different formats
• RR
• SR
• SDES
• BYE
• APP

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RTCP formats

• Reciever Report (RR)
• Sent by all participants that receive RTP packets
• Loss fraction Field
• Interarrival Jitter
• RFC 3550 6.4.1
• Sender Report (SR)
• Information about sent media for synchronization
  of several streams.
• Sent by participants that send RTP packets
• Sender Packet Count
• Octet/Byte count
• RTP timestamp

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RTCP formats

• Source Description (SDES)
• Consist of Items
• Canonical Name (CNAME), telephone nr, email etc.
• CNAME Persistent Identifier unlike SSRC
• Bye
• Indicates that the sender is leaving the session.
• Application-Defined RTCP Packets (APP)
• Application specific data

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Translators receiving RTCP packets

• Take apart compound packets
• Changing the encoding
• Forwards the SR packets
• Modify byte count
• Forwards other packets unchanged

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Translators receiving RTCP packets

• Combining Packets
• Forwards the SR packets
• Modify packet count
• Modify the RTP timestamp
• Forwards the RR packets
• Modify Packet Loss field
• Forward other packets unchanged

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Translators receiving RTCP packets

• SDES
• May filter away information (items) except CNAME
  when bandwidth is limited
• BYE
• Sent unchanged.
• APP
• Sent unchanged

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Mixers receiving RTCP packets

• Will NOT forward SR or RR packets
• Instead they will generate their own SR and RR
  packets
• SDES
• May filter away information (items) except CNAME
  when bandwidth is limited
• Extracted from compound packets to be inserted in
  new compound packets with new SR and RR.

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Security

• Encrypted packets
• Enterprises may want encrypted conferences
• Need to know the key to decode, read and modify
  information.
• Trusting the intermediate devices
• Who provides the service?
• Set up your own translator and mixer.

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Setup your own translators or mixer

• Reflex
• http//w2.alkit.se/reflex/
• GOSSIP
• www.swt.fh-mannheim.de/gossip/download.html

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Conclusion

• Provides complexity to the protocol but it has
  been shown that they are needed.
• Open source alternatives.
• As long as networks will suffer from low
  bandwidth they will remain.