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TRANSCRIPT
Welcome!
Today’s Webinar: Designing VoIP Services
with PIKA Building BlocksIrene CrosbyHead of MarketingPIKA Technologies
Your Webinar Leader
Yashar MoghanSenior Field Application EngineerPIKA Technologies
Familiarization with MShow
• Audio streaming• Sending us questions
Objectives
To present:• A technical overview of PIKA VoIP
building blocks, and• How to develop VoIP-enabled
applications using a PIKA platform
• 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
Agenda
• You have a basic technical understanding of IP / VoIP
• Ideally, you are familiar with PIKA MonteCarlo
Assumptions
Jitter Buffer
RTP
RTP 001010 RTP
0010101010
Codec
01001001010101001110101
1101110101
Echo
0010101010
Codec
RTP
RTP 001010 RTP
SIP IP
PSTN/POTS PIKA APPLICATION IP
General requirements for IP-PSTN connectivity
Essential building blocks in MonteCarlo 6.2
1. Readily supported IP codecsG.711, G.726
2. RTP Packet headers containing codec type and ordering info
3. Jitter bufferUses RTP information to deliver smoother /steady audio
4. Echo cancellation
All done on the DSP
Essential building blocks in MC6.2
1. 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
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
Essential building blocks in MC6.2
3. 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
Essential building blocks in MC6.2
4. 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
Essential building blocks in MC6.2
PIKA MonteCarlo 6.2 VoIP API: PK_VOIP_xxx
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
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;
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;
VoIP API – Encode – payloadType• PK_VOIP_EncodeSetParameters( )
Payload Type
Codec Payload TypeG.711 µ-law 0G.711 A-law 8G.726 2 / dynamic
For details see: www.ietf.org/rfc/rfc3551.txt
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]
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).
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
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
Related API – called before VoIP API
• PK_DSP_GetDeviceHandle
• PK_DSP_DEVICE_SeizePort
• PK_DSP_PORT_SetEventHandle• PK_CTBUS_FullDuplexConnect //Line Port
DSP Port
Related API – SeizePort (DSP)
• PK_DSP_DEVICE_SeizePort (TDeviceHandle hDsp, TResourceMask resMask );
Example:
resMask = PK_RTP|PK_G711|PK_ECHO_CANCELLATION
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
VoIP API – Decode – SetParameters• PK_VOIP_DecodeSetParameters (TResourceHandle hPort, TVOIPDecodeParameters *decodeParams);
typedef struct{TCodecType codecType;PK_U32 payloadType;PK_U32 initialLatencyInFrames; //default 3PK_BOOL dynamicJitterBufferEnabled; //TRUE or
FALSEPK_U32 jitterBufferPeriod; //default 640msPK_U32 fixedLatencyInFrames; //1-11
} TVOIPDecodeParameters;
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
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
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
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
VoIP API – Decode – SetParameters - ExampleTVOIPDecodeParameters 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 );
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
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)
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);
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 with
H.323, MS SIP, oSIP, Vovida• MonteCarlo 6.3 to provide an embedded
SIP stack as well
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
Setting up DSPs for VoIP using PikaSetup.exe
VoIP Test and Sample Applications
Test application
• PikaTest.exe• Part of MonteCarlo 6.2 installation, under PIKA Bin
folder• General multi-purpose application• Command-line based
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
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
SIP demo configuration
Telephony Switch
SIP Proxy
SIP Server 1
NIC
Daytona
SIP Stack + MC 6.2SIP Server 2
NIC
Daytona
SIP Stack + MC 6.2
Any phone
GrandStream VoIP phone
ext. 441 ext. 442
ext. 450
External line(591-0000)
PIKA WORLD
PSTN WORLD
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…
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 • [email protected]• Phone: +1-613-591-1555 x216
Thank youfor your time.