Welcome! Today’s Webinar: Designing VoIP Services with PIKA Building Blocks

1. Welcome! Today’s Webinar: Designing VoIP Services with PIKA Building Blocks • Irene Crosby • Head of Marketing • PIKA Technologies

2. Your Webinar Leader • Yashar Moghan • Senior Field Application Engineer • PIKA Technologies

3. Familiarization with MShow • Audio streaming • Sending us questions

4. Objectives To present: • A technical overview of PIKA VoIP building blocks, and • How to develop VoIP-enabled applications using a PIKA platform

5. Agenda • An overview of general requirements for IP-PSTN connectivity • VoIP building blocks in MonteCarlo 6.2 • G.711 and 726 codecs, RTP, jitter buffer, EC • PIKA VoIP and related API • Integration with VoIP signaling stacks • Hardware requirements: • PIKA MM cards plus any host NIC • Setup tool, test and sample applications

6. Assumptions • You have a basic technical understanding of IP / VoIP • Ideally, you are familiar with PIKA MonteCarlo

7. RTP RTP 001010 RTP 0010101010 RTP 001010 RTP RTP 0010101010 Jitter Buffer General requirements for IP-PSTN connectivity PSTN/POTS PIKA APPLICATION IP Codec 01001001010101001110101 IP SIP Echo 1101110101 Codec

8. Essential building blocks in MonteCarlo 6.2 • Readily supported IP codecs G.711, G.726 • RTP Packet headers containing codec type and ordering info • Jitter buffer Uses RTP information to deliver smoother /steady audio • Echo cancellation All done on the DSP

9. Essential building blocks in MC6.2 • IP codecs • Format of Tx/Rx compressed data/audio • Compressed data = Payload • G.711 and G.726 formats are offered readily • Other codecs would be considered as needed

10. Essential building blocks in MC6.2 • RTP (Real-Time Transport Protocol) • Serves as a mechanism to tell the receiving-end the format of the audio • Provides information to the receiving-end to put the packets in the correct order or skip missing ones • [RTP header + Payload] => RTP IP packet

11. Essential building blocks in MC6.2 • Jitter buffer mechanism • Why needed: VoIP packets do not necessarily arrive at their destination in correct order, or at all! • An adjustable buffer with parameters set in the API structure; i.e. size, dynamic vs. static • Takes advantage of RTP information associated with each packet to order them correctly before delivering the audio to the near-end listener

12. Essential building blocks in MC6.2 • Echo Cancellation • Due to inherent delay in VoIP systems, echo is more noticeable than in regular PSTN calls. It must be removed in most cases! • G.168 compliant • Tail lengths from 1 to 128ms • EC mask is simply added to resMask when seizing the RTP resource

13. PIKA MonteCarlo 6.2 VoIP API:PK_VOIP_xxx

14. VoIP API • PK_VOIP_EncodeSetParameters • PK_VOIP_EncodeGetBufferSize • PK_VOIP_EncodeAddBuffers • PK_VOIP_EncodeStart • PK_VOIP_EncodeStop • PK_VOIP_DecodeSetParameters • PK_VOIP_DecodeGetBufferSize • PK_VOIP_DecodeStart • PK_VOIP_DecodeAddBuffers • PK_VOIP_DecodeStop

15. VoIP API – Encode – SetParameters • PK_VOIP_EncodeSetParameters (TResourceHandle hPort,TVOIPEncodeParameters *encodeParams); typedef struct{ TCodecType codecType; PK_U32 payloadType; PK_U32 packetizationRate; <keep default 2> PK_BOOL vadEnabled; } TVOIPEncodeParameters;

16. VoIP API – Encode – codecType • PK_VOIP_EncodeSetParameters( ) Codec Types typedef enum{ PK_PCMU = 0, PK_PCMA = 1, PK_G726_16 = 4, PK_G726_24 = 5, PK_G726_32 = 6, PK_G726_40 = 7 } TCodecType;

17. VoIP API – Encode – payloadType • PK_VOIP_EncodeSetParameters( ) Payload Type Codec Payload Type G.711 µ-law 0 G.711 A-law 8 G.726 2 / dynamic For details see: www.ietf.org/rfc/rfc3551.txt

18. VoIP API – Encode – GetBufferSize • PK_VOIP_EncodeGetBufferSize (TResourceHandle hPort); • Must be called after SetParameters but before AddBuffer • Returns the recommended size of the buffer (in bytes) • e.g. if G.711, packetization 2: required RTP buffer size returned is 172 bytes (12 bytes RTP header + 2 x 80 bytes payload) • Used to allocate memory for the buffer (.lpData) • Used to set the length of the buffer (.dwBufferLength) [See Page 202 of Programmer’s Guide]

19. VoIP API – Encode – AddBuffer • PK_VOIP_EncodeAddBuffer (TResourceHandle hPort, TBufferHeader *pBuffer); • Called after EncodeGetBufferSize but before EncodeStart • The buffer passed-in must at least be as large as the buffer size returned by PK_VOIP_EncodeGetBufferSize • The buffer is returned to the application once it is filled (by RTP encoding, hport).

20. VoIP API – Encode – Start • PK_VOIP_EncodeStart (TResourceHandle hPort); • Called after EncodeAddBuffer • Starts RTP encoding process on DSP port, hPort • Encoded RTP packets are returned to application via installed ‘event handler’ (of hport) and indicated by event PK_EVENT_VOIP_ENCODE_RETURN_PACKET • The packet is now ready for transmission over the host network interface

21. VoIP API – Encode – Stop • PK_VOIP_EncodeStop (TResourceHandle hPort); • Called after Encodestop • Stops an RTP encoding operation on the DSP port specified by hPort • Used when the VoIP call terminates

22. Related API – called before VoIP API • PK_DSP_GetDeviceHandle • PK_DSP_DEVICE_SeizePort • PK_DSP_PORT_SetEventHandle • PK_CTBUS_FullDuplexConnect //Line Port  DSP Port

23. Related API – SeizePort (DSP) • PK_DSP_DEVICE_SeizePort (TDeviceHandle hDsp, TResourceMask resMask ); Example: resMask =PK_RTP|PK_G711|PK_ECHO_CANCELLATION

24. VoIP API • PK_VOIP_EncodeSetParameters • PK_VOIP_EncodeGetBufferSize • PK_VOIP_EncodeAddBuffers • PK_VOIP_EncodeStart • PK_VOIP_EncodeStop • PK_VOIP_DecodeSetParameters • PK_VOIP_DecodeGetBufferSize • PK_VOIP_DecodeStart • PK_VOIP_DecodeAddBuffers • PK_VOIP_DecodeStop

25. VoIP API – Decode – SetParameters • PK_VOIP_DecodeSetParameters (TResourceHandle hPort, TVOIPDecodeParameters *decodeParams); typedef struct{ TCodecType codecType; PK_U32 payloadType; PK_U32 initialLatencyInFrames; //default 3 PK_BOOL dynamicJitterBufferEnabled; //TRUE or FALSE PK_U32 jitterBufferPeriod; //default 640ms PK_U32 fixedLatencyInFrames; //1-11 } TVOIPDecodeParameters;

26. VoIP API – Decode – initialLatencyInFrames • PK_VOIP_DecodeSetParameters( ) initialLatencyInFrames: • Indicates number of frames that will be placed in the jitter buffer before starting the RTP decoder • Suggested/default value is 3 • 0 or 1 will have the RTP receiver start the decoder as soon as an RTP packet is received

27. VoIP API – Decode – dynamicJitterBufferEnabled • PK_VOIP_DecodeSetParameters( ) dynamicJitterBufferEnabled: • Set as PK_TRUE or PK_FALSE • RTP process will manage the number of frames to put in the jitter buffer by analyzing the packets received • After the jitterBufferPeriod is expired, the number of “initial frames” in the jitter buffer may change based on timing of the packets received

28. VoIP API – Decode – jitterBufferPeriod • PK_VOIP_DecodeSetParameters( ) jitterBufferPeriod: • Defines how often the RTP receiver manages the jitter buffer in number of 10 milliseconds • Default value is 640 ms

29. VoIP API – Decode – fixedLatencyInFrames • PK_VOIP_DecodeSetParameters( ) fixedLatencyInFrames: • Identifies the fixed number of frames to be stored in the jitter buffer • 1 is the minimum, 11 is the maximum value for this parameter • Applied when dynamicJitterBufferEnabled is set to PK_FALSE

30. VoIP API – Decode – SetParameters - Example TVOIPDecodeParameters VOIPDecodeParameters ; VOIPDecodeParameters.codecType = PK_PCMU; VOIPDecodeParameters.payloadType = 0; VOIPDecodeParameters.initialLatencyInFrames = PK_VOIP_INITIAL_LATENCY_DEFAULT; // 3 (frames) VOIPDecodeParameters.dynamicJitterBufferEnabled = PK_TRUE; VOIPDecodeParameters.jitterBufferPeriod = PK_VOIP_JITTER_BUFFER_PERIOD_DEFAULT; // 64 (640ms) VOIPDecodeParameters.fixedLatencyInFrames = 0; PK_VOIP_DecodeSetParameters ( hDSPPort, &VOIPDecodeParameters );

31. VoIP API – Decode – GetBufferSize • PK_VOIP_DecodeGetBufferSize( ) • Provided for consistency with Encode • Returned value is calculated based on the size of RTP header and the maximum allowed payload (200 ms) • Application may allocate smaller buffers if the worst case is known for the expected number of frames per RTP packet

32. Important notes: • DSP port seized for VoIP streaming can perform both encode and decode simultaneously; i.e. one DSP port per VoIP session • For encode: add ~15 buffers before calling PK_VOIP_EncodeStart • For decode: use of only one buffer is sufficient; to be added after calling PK_VOIP_DecodeStart (once filled with an incoming packet)

33. Related API – Echo Cancellation • PK_DSP_DEVICE_SeizePort(hDSPDevice , PK_ECHO_REFERENCE); • PK_CTBUS_HalfDuplexConnect(hDSPPort , hDSPPortER); • PK_EC_Initialize(hDSPPort , hDSPPortER , 12); //last parm Tail Length • PK_EC_Enable(hDSPPort);

34. Integration with VoIP signaling stacks • No restriction on use of any stack due to low-level, modular and flexible VoIP API • Sample integration with SIP is available • Customers have done integration withH.323, MS SIP, oSIP, Vovida • MonteCarlo 6.3 to provide an embedded SIP stack as well

35. PIKA board support and other hardware requirements • All PIKA board types support VoIP • PrimeNet MM (E1/T1), Daytona MM (LS/POTS), InLine MM (LS) • DSP-based, common to all boards • Host NIC used for IP connection

36. Setting up DSPs for VoIP using PikaSetup.exe

39. VoIP Test and Sample Applications

40. Test application • PikaTest.exe • Part of MonteCarlo 6.2 installation, under PIKA  Bin folder • General multi-purpose application • Command-line based

41. Test/sample applications • VoIP_Sample_DMM • Demonstrates use of VoIP API in a focused manner • No signaling stack integration • Easy setup, requires one DMM POTS only • VoIP audio streaming between two phones on DMM • Echo cancellation enabled • Available online under Downloads Sample Code

42. Test/sample applications • SIP Demo v0.5 • Much larger application, includes SIP integration • Able to register the system (i.e., a VoIP client) with a publicly available or private SIP proxy • Full ‘call’ support; including invite, trying and bye messages • Can place VoIP calls between two PIKA-based systems (VoIP clients) or to a SIP phone • Verified under numerous SIP proxies • Online under Downloads Sample Code

43. SIP Proxy SIP demo configuration ext. 450 PSTN WORLD GrandStream VoIP phone SIP Server 1 SIP Server 2 SIP Stack + MC 6.2 NIC SIP Stack + MC 6.2 NIC Daytona Daytona External line (591-0000) Telephony Switch ext. 441 ext. 442 PIKA WORLD Any phone

44. Thank You • If you are interested in taking a closer look, download our Programmer’s Guide: • http://www.pikatechnologies.com/downloads/software.htm • Additional HW info is available from: • http://www.pikatechnologies.com/downloads/hardware.htm • If you want to speak to the sales account manager in your region, a field application engineer, or technical support, the next slide has their contact information…

45. How to reach our people • Sales • Western USA – Brett Sumpter phone: +1-903-939-3711 • Eastern USA – Cheryl Farmer phone: +1-770-345-5944 • EMEA – Maarten Kronenburg phone: +31 76 5083 560 • Canada, Americas & Asia – Terry Atwood phone: +1-613-591-1555 x329 • Field Application Engineers • Yashar Moghan – phone: +1-613-591-1555 x415 • Cindy Xu – phone: +1-613-591-1555 x458 • Technical Support • support@pikatech.com • Phone: +1-613-591-1555 x216

46. Thank youfor your time.