Mobile VoIP Software for Android and iOS devices
Mobile VoIP software extends the developer’s toolset with Android and iPhone native SDK

Mobile VoIP software libraries provide native MVoIP solutions for iPhone & iPad and Android phones, tablets and other mobile devices. These libraries extend the developer’s toolset with our native software development kit (SDK) that includes VoIP stacks, industry standard voice codecs and our robust Speech Enhancement algorithms.

  • Android and iOS VoIP SDKs
  • Native library with API
  • Narrowband and wideband voice codecs available
  • Echo and noise solutions for superior speech
  • Full featured VoIP telephony

Request MVoIP and Mobile VoIP solutions brochure

Our extensive experience in the design, implementation, and configuration of vocal mobile, network, and telephony applications will help you select the right solution for your mobile VoIP application. VOCAL’s MVoIP source code is optimized for DSPs and ARM processors from TI, ADI and other leading vendors. Contact us for a demo or to discuss your mobile VoIP requirements.

Mobile VoIP Software

VOCAL Mobile VoIP software is optimized for performance on DSPs and ARM processors for  iOS/iPhone/iPad, Android and other mobile over IP devices and includes the following:

  • Native library interface
  • Mobile SIP stack with adaptive jitter control
  • VoIP codecs support narrowband and wideband audio
  • Speech enhancement with echo cancellation, noise reduction and dereverberation
  • Beamforming for microphone noise reduction
  • VoIP Software Modules

VOCAL Supported platforms include: Texas Instruments, Linux, Android, iOS and Mac Devices, Analog Devices, and others

Android VoIP SDK

VOCAL’s Android VoIP SDK software library is optimized to meet unique SIP calling requirements for mobile over IP app developers with support for Android voice codecs, acoustic echo cancellation, and noise reduction algorithms. These libraries are developed  in C/C++ for use with the Android SDK.


VOCAL’s iOS VoIP SDK software library is optimized for custom mobile SIP client application development for iOS devices and include voice codecs as well as acoustic echo and noise cancellation algorithms. These libraries are developed in C/C++ for use with the iOS SDK.

Native SIP Stack

VOCAL’s SDK implements a native SIP stack with  VoIP protocols, including SIP, SDP and RTP optimized for iOS, Android and other platforms. The native SIP stack provides access to the speech coders and other software modules and incorporates an adaptive jitter algorithm to further enhance voice quality. The robust jitter algorithm is exceptional at maintaining constant voice output to the listener while quickly and effectively adapting to packet loss and network delays.

Mobile VoIP Codecs

VOCAL Mobile VoIP libraries incorporate a selection of  voice and audio codecs  to customize applications for unique MVoIP requirements including a full range of ITU, GSM, Wideband, Android, iOS, and other industry standard voice codecs optimized for mobile platforms. Developers should evaluate the characteristics of each vocoder and its suitability for a particular application and/or platform. VOCAL has extensive experience with the selection and configuration of these voice codecs for optimal performance on ARM platforms.

Mobile VoIP Speech Enhancement

VOCAL Mobile VoIP libraries include speech enhancement for improved speech intelligibility with optimized echo cancellation, dereverberation, and  noise reduction algorithms. Echo cancellation reduces reflected copies of a direct path wave from the acoustic / mechanical coupling between speaker and microphone; dereverberation handles reflections from surfaces in an enclosed environment; while noise reduction addresses additive noise sources, e.g. wind noise. The combination of these techniques produces superior speech in MVoIP applications.

Our mobile VoIP libraries also include acoustic beamforming algorithms to actively focus on and capture the speaker’s voice. Mobile devices are increasingly used as hands-free, speakerphone-like systems where the speaker is often located in a room (or automobile) some distance from the microphone. Beamforming  takes advantage of newer mobile devices with multiple microphones to selectively capture the speaker’s voice while rejecting interferes and other sources of background noise.

Mobile VoIP Telephony Features


Network Protocols








DHCP Client







NAT/Firewall Support

Built-in Router

NAT Traversal

NAT Firewall

Port Forwarding

LAN Pass Through


Security Protocols














Voice CodecsG.711G.723.1 – 6.4, 5.3 kbpsG.726 – 16, 24, 32, 40 kbpsG.728




Voice Features

Voice Activity Detection (VAD)

Silence Suppression (DTX)

Comfort Noise Generation (CNG)

Packet Loss Concealment (PLC)

Dynamic Jitter Buffer (Adaptive)

Audio Codec Preferences

Dynamic Payload Negotiation

Codec Name Assignment

Configurable Frames per Packet

G.168 Line Echo Cancellation

128ms Tail Length

NLP Suppression

Double-Talk Detection

Fax Support

G.711 Fax Pass-Through

T.38 – Real-Time Fax Over IP

T.38 using UDP

T.38 using RTP

TelephonyCLASS FeaturesCall Waiting Enable/DisableCaller ID Display Enable/DisableCall Waiting Caller ID Enable/Disable

Blocked Call List for a Specified #

Distinctive Ring for a Specified #

Block/Unblock Caller ID

Block/Unblock Caller ID for One Call

Accept Priority Call of a Specified #

Busy Number Redial

Call Return (Call the Last Caller)

Deactivate/Activate Call Waiting

Call Forwarding

Speed Dial (8 + 20 Numbers)

Block Anonymous Calls

Do Not Disturb

Call Transfer

3-way Conference Calling


Call Hold

Call Waiting/Flash

Flash Hook Timer

Delay Disconnect

Hot Line and Warm Line Calling

Call Blocking with Toll Restriction

Caller ID Generation

Call Waiting Caller ID with Name/#

Distinctive Ringing

Distinctive Call Waiting

TTY / TDD Support

MWI – Tone and Visual


Polarity Control

Call Progress TonesProgrammable Tone Generation PatternsFour Tones, Four On/Off Time Pairs

  • Initial Dial Tone
  • Secondary Dial Tone
  • Stuttered Dial Tone
  • Message Wait Dial Tone
  • Call Forward Dial Tone
  • Pre-Ringback Dial Tone
  • Ring Back Tone
  • Call Waiting Tone
  • Call Holding Tone
  • Call Disconnect Tone
  • Call Conference Tone
  • Busy Tone
  • Reorder Tone (Network/Fast Busy)
  • Off Hook Warning Tone (Howler Tone)
  • SIT Tones 1 to 4
  • Prompt Tone
  • Confirm Tone
  • Input Error Tone
  • Number Error Tone

Ringing Patterns

Programmable Ring Patterns

Four On/Off Time Pairs

  • Default Ring
  • Hold Rering
  • Call Back Ring
  • Call Back Ring Splash
  • Call Forward Ring Splash
  • Message Waiting Ring Splash
  • 8 Distinctive Ring Patterns

Distinctive Call Waiting

Programmable Tone Generation

Four Tones, Four On/Off Time Pairs

8 Distinctive Call Waiting Patterns

SIPv2 – Session Initiation Protocol (RFC 3261, 3262, 3263, 3264)

SDP – Session Description Protocol (RFC 4566)

RTP – Real-Time Protocol (RFC 3550, 3551)

RTCP – Real-Time Control Protocol (RFC 3550)

RFC 4733 X-NSE Tone Events for SIP/RTP

RFC 4733 AVT Tone Events for SIP/RTP

STUN – Simple Traversal of UDP over NATs (RFC 3789)

SIPS – SIP Secure using TLS (RFC 3261)

SRTP – Secure Real-time Transport Protocol (RFC 3711, 4568)

MKI – Master Key Identifier (part of RFC 3711)

AES – Advanced Encryption Standard – supports 128/195/256 bit keys

HMAC – Authentication

IPv4 – Internet Protocol Version 4 (RFC 791)

TCP – Transmission Control Protocol (RFC 793)

UDP – User Datagram Protocol (RFC 768)

ICMP – Internet Control Message Protocol (RFC 792)

RARP – Reverse Address Resolution Protocol (RFC 903)

ARP – Address Resolution Protocol (RFC 826)

DNS- Domain Name Server

DHCP Client – Dynamic Host Control Protocol (RFC 2131)

NTP – Network Time Protocol (RFC 1305)

SNTP – Simple Network Time Protocol (RFC 2030)

HTTP – HyperText Transfer Protocol

TFTP – Trivial File Transfer Protocol (RFC 1350)

PPPoE – Point to Point Protocol over Ethernet (RFC 2516)

G.711 – Pulse Code Modulation

G.722 – Wideband ADPCM

G.722.1 – 24k and 32k bps 7kHz Wideband

G.722.2 – GSM-AMR-WB

G.723.1 – 6.4 and 5.3 kbps ACELP/MP-MLQ

G.726 – 16, 24, 32 and 40 kbps ADPCM

G.728 – 16 kbps LD-CELP

G.729 – 8 kbps CS-ACELP

G.729A – 8 kbps CS-ACELP Low Complexity

G.729B – Silence Detection/Comfort Noise Generation

G.729D – 6.4 kbps CS-ACELP

G.729E – 11.8 kbps CS-ACELP

GSM-FR – GSM 06.10 Full Rate Vocoder

GSM-AMR NB – GSM 06.90 Adaptive Multi-Rate

GSM-AMR WB – Wideband Adaptive Multi-Rate

iLBC – Internet Low Bitrate Codec

OPUS – 16KHz SILK, 22/24KHz CELT

Speex – 8 kbps CELP

MELPe – 2400/1200/600 bps Codec

TSVCIS – Tactical Secure Voice (Wideband MELPe)

LPC10 and CVSD – Legacy Voice Codecs

Q.24 DTMF Generation with Zero Crossing Cutoff

Q.24 DTMF Detection exceeding Bellcore Specifications

Configurable Tone Generation for 4 Sets of Frequencies and 4 Sets of On/Off Cadence

Programmable Precise Tone Detectors

G.168 Line Echo Cancellation

16/32/64 ms Echo Length

Nonlinear Echo Suppression (ERL greater than 28 dB for f = 300 to 3400 Hz)

Double-Talk Detection

Full Duplex Speakerphone

Narrow and Wideband Operation (8Khz and 16KHz)

Adjustable Tail Length (128 ms typical, 256 ms max)

Nonlinear Echo Processing with Comfort Noise Generation

Full Duplex Operation with Noise Reduction

Double-Talk detection, Low Divergence during Double Talk

Dual/Multi-Microphone Adaptive Noise Cancellation

Single Channel Noise Reduction

Active Noise Cancellation to identify and remove repetitive noise signals

Frequency Domain Noise Reduction

18dB Noise Reduction (typical)

Approximate 20msec Delay

Audio Beamforming (four or more mics)

Audio Null Forming

Direction of Arrival Estimation

Audio Beam Steering

Automatic Gain Control

Voice Compressor

Multiband Equalizer

Automatic Delay Estimation

Battle Field VOX

Voice Activity Detection

Noise Gating

Wind Noise Reduction

Click Noise Removal

VAD/DTX/CNG as per SCIP 210 Appendix B