Helsinki University of Technology Department of Electrical and Communications Engineering

1. Helsinki University of Technology Department of Electrical and Communications Engineering Jarkko Kneckt point to point and point to multi point calls over IP Helsinki 27.11.2001 Supervisor: Raimo Kantola Instructor: Heikki Salovuori MSc.

2. AGENDA • Different types of call services • Differences between simplex (streaming) and duplex calls • Role of server in simplex calls • Used protocols • General introduction to VOIP protocols • Discussion on what simplex connections need for signaling protocol • Introduction to our choise for simplex call signaling • Voip client implementation • General operation of VOIP client • General architecture of simplex call type VOIP client

3. Call services

4. Why have simplex calls 1 1.No quality of service in internet • Streaming call is send only one direction. Receiver cannot speak at the same time as speaker is speaking. => Receiver can use bigger jitter buffer to buffer packet sending speed variation than used in normal VOIP applications. 2. New way to communicate • Message to one receiver (or many receivers) at the time. From previous same kind applications average duration of one simplex call is 7 seconds. This causes special needs to be able to start call fast, setup should take 0,5 – 1 s.

5. Why have simplex calls 2 3.Streaming call is easy to copy several receiver Servers in picture areused only by this application. Servers forward this application packets. IP network

6. New problems in simplex call handling Problems in simplex call system handling: (These problems define the role for ”server system” when simplex call mode is used 1. Which call should be played? How to minimize traffic , especially air traffic to phone? 2. Should outgoing call be ended? Should starting call be dropped?

7. New problems in simplex call handling 3. How to keep a record who are using the service? 4. How to specify which calls you want to listen? 5. How to specify rights for calls ? Who can speak? 6. Usability. How to use service so that it is easy and simple to use?

8. protocols in VOIP SIP and rtp handling are in the scope of this work

9. Functions of RTP • RTP Real Time Protocol • RTP is protocol, which provides timing information to packets. • RTP protocol does not include jitter (= time buffering for variation of packet arrival times) buffering function, but jitter buffering can be made according to information in RTP headers. Jitter buffering is own application. • Normally used to add time information to packets from sender to receiver. Also used to specify realtime connection and codec (audio / video) which has created the sent data.

10. Streaming special needs for signalling Problem in point to multipoint call. call should start ride away we don’t have Normally SIP or H.323 is used for VOIP call signalling. Now we are not interested to get acknowledge from every called party in point to multipoint call. Sip is not appropriate signalling protocol because according to protocol we must get acknowledgement from receiver. However SIP is very easy to adjust the needs for signaling. Sip can be used as signaling protocol in many different applications. Acknowledgement in point to multipoint calls would introduce unnecessary delays when ack message is waited. Acknowledged setup messages

11. How to speed up setup time ? • Usage of acknowledgement in point to multipoint calls setup messages • would introduce unnecessary delays when ack message is waited. • On the other hand if we receive some acks but not all should we wait for • all acks to arrive or should the call be started right away? • question is : how to set limits when point to multipoint call can be started? • Better solution for setup messages is to define a time after call is started. • No acknowledgements etc. is send. • Normally in VOIP calls in call setup phase also the codec is chosen. If we could agree on • all issues that are not directly depending on call setup, (used codec, receiver(s)) • so that actual setup message contains only relevant information. • SIP or H.323 cannot be used for this kind of signaling. New signaling protocol is needed: One solution is to transfer signaling information on top of RTP. • Also completely new protocol could be created on top of UDP.

12. SIP usage • SIP (Session Initialization protocol) contains ready made signaling messages without payload. Payload can be add with SDP Session Description Protocol. Only the format of payload is defined, not sent data. • The role of SIP is a bit different than in normal VOIP solutions • SIP is used for: • sending log on /log off messages to service (SIP : REGISTER) • defining which calls are received • defining in point to multipoint calls who are callees

13. Pros and cons in rtp usage in signalling Positive Negative Originally planned for real time data transfer Works on top of UDP => no connection oriented benefits. No automatic packet loss detection or resend RTP is well specsed, ready designed interface RTP is used in almost all VOIP solutions. Easier to use ready standardized protocol than try to standardize a completely new protocol. RTP header needs only new payload type for signaling

14. RTP header 0 31 V P X CC M Payload type sequence number 12 Bytes Symbol defination and binary value if standard lenght in bits • V Version. Identifies the version of rtp 2 • P Padding. When set, the packet contains one or more additional padding octetsat the, end which are not part of the payload 1 • X Extension bit. When set the fixed header is followed header is followed by exactly one extension 1 • CC CSRC count. Number of CRSC identifiers that follow fixed header 4 • M Marker bit. The interpretation of the marker is defined by a profile. It is intented to allow significant events such as frame boundaries to be marked in the packet stream 1 • PT Payload type. Identifies the format of the RTP payload and determines its interoperability by the application. A profile spacifies a default static mapping of payload type codes to payload formats. Additional payload type codes may be defined dynamically through non-RTP means 7 • sequence number increased by one for each RTP packet sent, and may be used by the default to detect packet loss and to restore packet sequence 16 • timestamp Reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations. 32 • SSRC identifies the synchronization source. This identifier is chosebn randomly, with the intent that two synchronization sources will have same SSRC identifier 32 Timestamp SSRC

15. RTP header usage in signaling • RTP header can be used for signaling purposes by defining a payload type for signaling. This way application knows that message contains signaling. • All the other fields can remain the same. • Application must build retransmit and confirmation mechanism in message charts. • RTP signaling is used only for setup and termination messages. V P X CC M Payload type sequence number Timestamp SSRC

16. RTP signaling example 1 Point to multipoint call: All signaling messages are sent 3 times to avoid packet loss errors Call setup 1. Leading RTP packet 1. Leading RTP packet 1-3 setup message 2. Leading RTP packet 2. Leading RTP packet 3. Leading RTP packet 3. Leading RTP packet Call ongoing 4 Audio packets 4.Audio packets of the call 4.Audio packets of the call Call termination 5. Trailing RTP packet . 5. Trailing RTP packet 6. Trailing RTP packet 6. Trailing RTP packet 5-7 Termination messages 7. Trailing RTP packet 7. Trailing RTP packet Client 1 Server Client 2

17. RTP Signaling example 2 Point to point call All signaling messages are sent 3 times to avoid packet loss errors When we receive ACK we can stop sending leading RTP Call setup 1. Leading RTP packet 1-2 setup message 1. Leading RTP packet 2. Leading RTP packet 2. Leading RTP packet 3-5 ACK messages to setup. 3. RTP ACK 3. RTP ACK 4. RTP ACK 4. RTP ACK 5. RTP ACK 5. RTP ACK 6 Audio packets Call ongoing 6.Audio packets of the call 6.Audio packets of the call Call termination 7. Trailing RTP packet 7-9 Termination messages 7. Trailing RTP packet 8. Trailing RTP packet 8. Trailing RTP packet . 9. Trailing RTP packet . 9. Trailing RTP packet Client 1 Server Client 2

18. General requiremnets for client • Work share between client and server: • (see slides 6-7 for serverside general problems of simplex calls) • Client proposes new calls server decides can client start a call • Server looks that client receives only one call at the time • Client takes care of voice handling. Server only forwards voice • packets to clients. • My implementation (as a part of master thesis work): • Client working on linux in laptop, connected with Wlan • and IPv6 to internet. • Must have support for RTP signaling, SIP signaling and graphical user • interface.

19. client architecture

20. RTP packet handling flow chart when rtp contains signaling and audio packets Flow chart for call state logic shown in previous slide.

21. Flow chart of audio handling in VOIP Audio Audio Audio device Audio device Loudspeaker Microphone D/A conversion A/D conversion Audio device driver Audio device driver Media subsystem Media subsystem Decoding Encoding Deframing Framing Jitter buffer RTP packetization RTP depacketization UDP / IP packetization UDP / IP depacketization Network device driver Network device driver Network device Network device Physical transmission server in internet ornothing Physical transmission internet internet