Eyeball AnyBandwidth Engine V8.0 Developer’s Guide

Copyright © 2002-2015 Eyeball Networks Inc. Patented and patents pending. All rights reserved.
Eyeball AnyBandwidth™ v8.0
Developer's Guide
Last Modified: November 2014

1. ABE Introduction
Introduction
The Eyeball AnyBandwidth™ Engine (ABE) enables audio-video communications among multiple clients
and/or servers having heterogeneous processor capacities and network connections, and/or network
connections with dynamically changing properties.
AnyBandwidth provides optimized video quality (such as frame rate, picture resolution, and sharpness) to
each user based on the available network bandwidth and processing power.
AnyBandwidth Engine contains Eyeball’s patented AnyBandwidth Technology, and is found in several
Eyeball products including Eyeball Messenger and Eyeball Messenger SDK. Messenger SDK utilizes the
SIP signaling protocol, but this does not preclude AnyBandwidth Engine from utilizing other signaling
protocols based on application demand.
Eyeball AnyBandwidth Engine offers the following features:
 Guaranteed voice and video quality
 Packet loss concealment
 Flexible voice processing (AEC, AGC, CNG, NS, VAD) for hands-free handsets and tablets
 Plug-and-play voice codecs including G.711, G.729, GSM, iLBC, Siren, Speex
 Plug-and-play video codecs including EyeStream, H.263, H.264

Figure 1: AnyBandwidth Engine block diagram.

2. ABE Supported Platforms
Supported Platforms
 2.1. ABE: Microsoft Windows
 2.2. ABE: Android
 2.3. ABE: iOS

2.1. ABE: Microsoft Windows
Microsoft Windows
System Requirements
Operating System:
XP, Vista, Win7
Hardware requirements:
Audio device, camera for video enabled calls
Developer Platforms
Programming Languages:
C/C++, C#, HTML and Javascript
Developer tools:
Visual Studio 2005 or later
Other software:
DirectX 8.1 and .Net Framework 1.1 or later versions

Programming Conventions
In this document, the phrase current user refers to the local user, as opposed to the remote user.
SIP references (including class or object names) don’t infer exclusive use of the SIP protocol.
We use the following conventions to define a variable’s data type:
 Variable name starting with ‘n’ such as nLine refers to an Integer data type
 Variable name starting with ‘s’ such as sUserID refers to a String data type
 Variable name starting with ‘b’ such as bOutgoingCall refers to a Boolean data type
 Variable name starting with ‘a’ such as aResponseInfo refers to a VB array data type. This is a
one-dimensional array. Storing two-dimensional information is simply done by concatenating rows
to each other, forming a one-dimensional array.
All strings are case-sensitive unless specified otherwise.
The calling convention for properties follows this format:
// Select a line
SipCommCtl.SelectedLine = 1;
// Retrieve the currently selected line
int nSelectedLine = SipCommCtl.SelectedLine;

2.2. ABE: Android
Android
System Requirements
Operating System:
Android 2.3 or later
Android tablet or Android phone
Developer Platforms
Java
Developer tools:
Android SDK 2.3.3 or later
Please see Programming Conventions section for Microsoft Windows.

The methods and properties are static and implemented in the JNIWrapper.
Unless otherwise specified, the calling convention for properties follows this format:
// Select a line
JNIWrapper.SipCommPutSelectedLine(1);
int nSelectedLine = JNIWrapper.SipCommGetSelectedLine();
For methods which return a VB array on Windows, return a string array to Android using this format:
String[] stats = JNIWrapper.SipCommGetAudioReceiverStat();
For notification events which provide elements in a VB array on Windows, the elements are provided as
arguments on Android using this format:
OnVideoSizeChange(long hWnd, int iWidth, int iHeight){}

2.3. ABE: iOS
iOS
System Requirements
Operating System:
iOS 5.0 or later
Phone or iPad
Developer Platforms
C/C++, Objective C/C++
Developer tools:
Xcode
Please see Programming Conventions section for Microsoft Windows.

The methods and properties are defined in class CSipComm in file MediSession.h. To access these
methods, an object of CSipComm needs to be created. Please see Section 3.1. Creating SIPComm
objects.
Unless otherwise specified, the calling convention for properties follows this format:
// Select a line
commCtrl->sipComm->put_SelectedLine(1);
int nSelectedLine;
commCtrl->sipComm->get_SelectedLine(&nSelectedLine);
For methods which return a string on Windows, the string must be passed by reference to iOS using this
format:
string sDisplayName;
commCtrl->sipComm->GetDisplayName(&sDisplayName);
For methods which return a VB array on Windows, a vector must be passed by reference to iOS using
this format:
vector<string> stats;
commCtrl->sipComm->GetAudioReceiverStat(&stats);
For notification events which provide elements in a VB array on Windows, the elements are provided as
arguments on iOS using this format:
OnVideoSizeChange(long hWnd, int iWidth, int iHeight){}

3. Using AnyBandwidth Engine SDK
Using AnyBandwidth Engine SDK
AnyBandwidth Engine SDK consists of libraries of ActiveX controls (Windows)/Java (Android)/C++ (iOS)
that provide programmers with a high-level interface to the main features and functions of the
AnyBandwidth Engine.
There are controls available to display and handle video windows (for video calls, used together with the
SIP Communicator control), and audio device detection.
In this section, we present AnyBandwidth Engine SDK from a web-programmer’s point of view. However,
programmers using other languages such as C++ and Visual Basic will also get a clear understanding of
the supported features and functionalities.
 3.1. ABE: Creating SIPComm Objects
 3.2. ABE: Features and Functions
 3.3. ABE: Handling Event Notifications
 3.4. ABE: Properties
 3.5. ABE: Methods
 3.6. ABE: Notification Events
 3.7. ABE: Error Handling

3.1. ABE: Creating SIPComm Objects
Creating SIPComm Objects
Windows:
The following HTML code shows how to embed the ActiveX control objects into a web page:
<OBJECT
id="SipCommCtl"
classid="CLSID: 968E1865-05A8-41dd-95B5-7D45B9701A57">
</OBJECT>
<OBJECT
id="VideoWindowCtl"
classid="CLSID: 0504639F-FBD4-4272-B232-AB9B21305618">
<param name=”IsWndless” value=true>
</OBJECT>
Android:
public class Messenger extends Activity implements SipEventHandler{
public void onCreate(Bundle savedInstanceState){
JNIWrapper.AudioInit((Activity)this);
JNIWrapper.SetSipEventHandler(this);
JNIWrapper.SipCommInit();
}
}

iOS:
iOS:
class CommCtrl : public CSipCommEventHandler {
public:
CSipComm *sipComm;
CommCtrl(){
sipComm = new CSipComm(this);
}
CommCtrl *commCtrl = new CommCtrl();

3.2. ABE: Features and Functions
Features and Functions
Some of the main features and functions of AnyBandwidth Engine SDK are described in the following
pages:
 3.2.1. ABE: SIP Communicator Control
 3.2.2. ABE: Call User

3.2.1. ABE: SIP Communicator Control
SIP Communicator Control
This control supports peer-to-peer audio/video data-transport. AnyBandwidth Engine SDK supports
multiple-lines (similar to multi-line PBX phones), enabling the following features:
 Identifying each call using a separate line while in multiple concurrent calls
 Choosing a specific available line to make the next call

3.2.2. ABE: Call User
Call User
How to make a video call
Caller
// Get the AnyFirewallEngine instance from the SIP component
CAnyFirewallEngine *pAFEngine;
GetAFEngine((void **)&pAFEngine);
// Create audio, video RTP/RTCP channels and the session
int iAudioRTPChannel = pAFEngine->Create(AF_CHANNEL_RTP);
int iAudioRTCPChannel = pAFEngine->Create(AF_CHANNEL_RTCP);
int iVideoRTPChannel = pAFEngine->Create(AF_CHANNEL_RTP);
int iVideoRTCPChannel = pAFEngine->Create(AF_CHANNEL_RTCP);
int iSession = pAFEngine->CreateSession(sMediaStream);
// Create offer SDP
string sSDP = pAFEngine->MakeOffer(iSession);
// Create the call message
// Send the call message
pAFEngine->SendTo(iServerChannel, sCallMessage, iCallMessageLength, …);
Callee
// Upon receiption of call request and accept the call
// Create audio, video RTP/RTCP channels and the session
// Create answer SDP
string sSDP = pAFEngine->MakeAnswer(iSession);
// Create the call response message
// Send the call response message

Caller
// Upon receiption of call accept message from callee
// Negotiate audio video payload
// Grab a line
int iLine = GrabLine(-1);
// Open the session
OpenSession(iLine, chAudioPayload, chVideoPayload, iAudioRTPChannel,
iAudioRTCPChannel, iVideoRTPChannel, iVideoRTCPChannel, true);
// Send an acknowledgement message to the callee
Callee
// Upon receiption of the acknowledgement message from caller
// Grab a line
// Open the session
End call
// Send end call message
// Close the session
CloseSession(iLine, chAudioPayload, chVideoPayload);
// Release the line
ReleaseLine(iLine);
How to make an audio call
Caller
// Get the AnyFirewallEngine instance from the SIP component
CAnyFirewallEngine *pAFEngine;
GetAFEngine((void **)&pAFEngine);
// Create audio RTP/RTCP channels and the session
int iAudioRTPChannel = pAFEngine->Create(AF_CHANNEL_RTP);
int iAudioRTCPChannel = pAFEngine->Create(AF_CHANNEL_RTCP);
// Create offer SDP
string sSDP = pAFEngine->MakeOffer(iSession);

Callee
// Create audio RTP/RTCP channels and the session
Caller
// Negotiate audio payload
// Grab a line
// Open the session
OpenSession(iLine, chAudioPayload, 127, iAudioRTPChannel, iAudioRTCPChannel, -1, -1,
true);
Callee
// Grab a line
// Open the session
End call
CloseSession(iLine, chAudioPayload, 127);
// Release the line
ReleaseLine(iLine);
How to add video to an audio call
Caller
// Create video RTP/RTCP channels and the session
int iVideoRTPChannel = pAFEngine->Create(AF_CHANNEL_RTP);

int iVideoRTCPChannel = pAFEngine->Create(AF_CHANNEL_RTCP);
// Modify the existing session
// Create offer SDP
pAFEngine->ModifySession(iSession, sMediaStream);
Callee
// Create video RTP/RTCP channels and modify the session
Caller
// Negotiate audio payload
// Grab a line
// Open the session
OpenSession(iLine, chAudioPayload, chVideoPayload, iAudioRTPChannel,
iAudioRTCPChannel, iVideoRTPChannel, iVideoRTCPChannel, true);
Callee
// Grab a line
// Open the session
End call
CloseSession(iLine, chAudioPayload, chVideoPayload);
// Release the line
ReleaseLine(iLine);

Send DTMF
Sending DTMF tones is required for PBX calls and Touch Tone services. AnyBandwidth Engine SDK
implements RFC 2833 with RTP payload type 101. In addition, DTMF using SIP INFO or in-band can be
used (please see DTMFMode in Properties section for additional details).
To send DTMF tones:
Windows:
SipCommCtl.SendDTMF(5);
Android:
JNIWrapper.SipCommSendDTMF(5);
iOS:
commCtrl->sipComm->SendDTMF(“5”);
The digit “5” is sent as a DTMF signal.

3.3. ABE: Handling Event Notifications
Handling Event Notifications
The applications send several event notifications to the application running on AnyBandwidth Engine
SDK. The application needs to handle these events as necessary. For example, SIP control will send
event notifications such as:
Video size change at the remote end
Windows:
<SCRIPT language=JScript for= SipCommCtl event= OnVideoSizeChange(v)>
return SipCommCtl_OnVideoSizeChange(v)
</SCRIPT>
function SipCommCtl_ OnVideoSizeChange(v)
{
var aInfo = v.toArray();
var nWidth = aInfo[1];
var nHeight = aInfo[2];
}
Android:
void OnVideoSizeChange(long hWnd, int iWidth, int iHeight)
{
//Handle event
}
iOS:
void OnVideoSizeChange(long hWnd, int iWidth, int iHeight)
{
//Handle event
}

3.4. ABE: Properties
Properties
AudioCodecs
This sets or retrieves preferred audio codecs.
Currently, Polycom® Siren™ (G722.1C, 24kHz, 48kHz), GSM, G.711 (A-law, µ-law), G.729 Annex A,
iLBC, and Speex codecs are supported. On Android and iOS G.711, G729 and Speex (wideband) are
supported. Codecs are described with space-delimited strings. This value can be changed during a call.
The following codec strings are supported: “SIREN24”, “SIREN48”, “SPEEX”, “SPEEX-WB”, “ILBC”,
“GSM”, “PCMU”, “PCMA”, and “G729”.
When acting as conference host, it is possible to add conference participants thus supporting different
codecs in a single conference. This is useful when adding participants from PSTN gateways which
usually only support a small selection of codecs.
DTMFMode
This sets or retrieves whether the SendDTMF method sends DTMF using the RTP payload (0), SIP INFO
method (1), or inband DTMF (2). The default value is 0.
In-band DTMF is designed to work only with high bit rate codecs, such as G711 and G722. For low bit
rate codecs such as G729, RTP payload or SIP INFO should be used instead of inband.
EnableAGC

This sets or retrieves whether or not Auto Gain Control (AGC) is used for the microphone input signal.
This is a Boolean variable and its default value is false. This value can be changed during a call.
EnableDenoise
This sets or retrieves whether or not noise is removed from the microphone input signal. This is a Boolean
value and its default value is true. This value can be changed during a call.
This property should be enabled for better echo cancellation.
EnableEchoCancellation
This sets or retrieves whether or not echo cancellation is enabled. This is a Boolean property and its
default value is true. This value can be changed during a call.
For better echo cancellation, EnableDenoise property must be set to true.
EnableIceSupport
This sets or retrieves whether or not ICE candidates are used in invite for firewall traversal. ICE stands for
Interactive Connectivity Establishment. This is a Boolean property and its default value is true.
EnableRelaySupport
This sets or retrieves whether or not relayed candidates are used in invite for firewall traversal.
AnyBandwidth Engine SDK will use a TURN server to obtain these candidates. This is a Boolean property
and its default value is true.
EnableSrtp
This enables or disables Secure RTP one-to-one calls. This property does not support conferencing. This
is a Boolean variable and its default value is false. It can only be set before making a call. The caller has

to enable SRTP, but the callee must support SRTP as well for the call to be secure, although it does not
need to call this property.
EnableStunSupport
This sets or retrieves whether or not server reflexive candidates are used in invite for firewall traversal.
AnyBandwidth Engine SDK will use a STUN server to obtain these candidates. This is a Boolean property
and its default value is true.
HoldLine
This holds or un-holds SelectedLine, or retrieves whether or not SelectedLine is being held. The default
value of this Boolean property is false.
ImageSize
This sets the image size to be captured, or retrieves the captured image size as an integer. The default
value is 0. This property can be set at any time. OnVideoSizeChange event is fired when the image size
is changed.
The possible values are:
Windows:
0:
Small image size (176 x 144, or 192 x 144, if supported by the device)
1:
Medium image size (352 x 288, or 320 x 240, if supported by the device)
2:

Large image size (640 x 480 if supported by the device)
3:
720p (HD) image size (1280 x 720 if supported by the device)
Android:
0:
Small image size (176 x 144)
2:
Medium image size (320 x 240)
3:
Large image size (352 x 288)
5:
VGA image size (640 x 480)
iOS:
1:
Small image size (192 x 144)
2:
Medium image size (320 x 240)
5:
VGA image size (352 x 288)

LineVolume
This sets or retrieves the volume of a SelectedLine. The value ranges from 1 (silence) to 100 (full
volume). The default value of this property is 100.
MuteReceiver
This mutes or un-mutes inbound audio on SelectedLine, or retrieves whether or not inbound audio on
SelectedLine is muted. The default value of this Boolean property is false.
MuteSender
This mutes or un-mutes outbound audio, or retrieves whether or not outbound audio is muted. The default
value of this Boolean property is false.
PauseReceiver
This pauses or un-pauses inbound video on SelectedLine, or retrieves whether or not inbound video on
SelectedLine is paused. The default value of this Boolean property is false.
PauseSender
This pauses or un-pauses outbound video, or retrieves whether or not outbound video is paused. The
default value of this Boolean property is false.
SelectedLine
This sets a line to be selected or retrieves a line that is currently being selected as an integer. Some
properties and methods perform operations on the line specified by SelectedLine.

VideoCodecs
This sets or retrieves preferred video codecs. Currently, EyeStream, H.263, and H264 codecs are
supported. On Android and iOS only H264 is supported. Codecs are described with space-delimited
strings. This value can be changed while in a call. The possible parameters are: “Eyestream”, “H263”,
“H263-1998” (H263+), and “H264”. When attempting an audio/video call to a client that does not support
the codecs selected in AnyBandwidth Engine SDK, the call will be completed as an audio only call.
In a conference, only one codec is supported. It is not possible to add participants that do not support the
video codec used by the existing conference participants.

3.5. ABE: Methods
Methods
AttachVideoWindow(nLine, nWnd)
This attaches a given video window to display video.
nLine:
This is the line indicating a call for video source. If nLine is -1, window is used to display the preview
video for the current user.
nWnd:
This is the window handle of the video window.
iOS:
AttachVideoWindow(int iLine, void *pWnd)
pWnd:
Reference to an UIView for camera preview surface, and to an UIImageView for remote video surface.
This is not applicable to Android environment.
AttachVideoWindowEx(nLine, nIndex, nWnd)
This attaches a given video window to the display video received from the specified line. A separate
instance of VideoWindowCtl is needed to display each video window.
nLine:

This is the line indicating a call for video source. If nLine is -1, window is used to display the preview
video for the current user.
nIndex:
This is the user index in the conference. It is 0 if not in a conference.
nWnd:
This is the window handle of the video window. This is returned by the GetVideoWindow method of
VideoWindowCtl.
This is not applicable to Android or iOS environments.
AudioInit(activity)
This initializes the audio manager.
activity:
An object of the Activity class
Please check the sample application code.
This is applicable to the Android environment only.
CloseSession(nLine, chAudioPayload,
chVideoPayload)
Closes audio, video in a session.
nLine:
The line
chAudioPayload:
Audio payload type in character format
chVideoPayload:

Video payload type in character format
GetAudioReceiverStat()
This returns audio receiver statistics of the currently selected line as a VB array of eight elements on
Windows, array of string on Android, and vector of string on iOS.
Element Description
Element 1 (Codec, String): Audio codec
Element 2 (Bit rate, Float): Audio receiving bit rate
Element 3 (Buffer size, Integer): Audio receive buffer size in ms
Element 4 (Audio resynchronization count,
Integer):
Number of times the audio receive buffer is reset and re-
synchronized
Element 5 (Packets received, Integer): Number of packets received
Element 6 (Packets lost, Integer): Number of packets lost
Element 7 (Packets late, Integer): Number of late packets
Element 8 (Loss rate, Float): Packet loss rate
GetAFEngine(void **pAFEngine)
Get reference of the instance of the AnyFirewallEngine used in the SIP component. This
AnyFirewallEngine instance is created and destroyed in the component itself.
pAFEngine:
Reference to CAnyFirewallEngine.
GetAudioSenderStat()
This returns audio sender statistics of the currently selected line as a VB array of three elements on
windows, array of string on Android and vector of string on iOS.
Element Description
Element 1 (Codec, String): Audio codec
Element 2 (Bit rate, Float): Audio receiving bit rate
Element 3 (Packets sent, Integer): Number of packets sent

GetFirewallStatus(nLine)
This returns the status of the firewall traversal on the call line specified by nLine.
Possible values:
“Unknown”:
Firewall traversal status is unknown
“Peer-to-peer”:
Firewall traversal succeeded and call completed peer-to-peer
“UDP Relay”:
Firewall traversal succeeded and call completed using UDP relay
“TCP Relay”:
Firewall traversal succeeded and call completed using TCP relay
“HTTP Relay”:
Firewall traversal succeeded and call completed using HTTP relay
“Fail”:
Firewall traversal failed
nLine:
The line indicating a call
GetVideoReceiverStat()
This returns video receiver statistics of the currently selected line (see SipCommCtl.SelectedLine) as a
VB array of eight elements.
Element Description

Element 1 (Codec, String): Video codec
Element 2 (Bit rate, Float): Video receiving bit rate
Element 3 (Frame rate, Float): Video receiver frame rate
Element 4 ((Frames received, Integer): Number of frames received
Element 5 (Packets received, Integer): Number of packets received
Element 6 (Packets lost, Integer): Number of packets lost
Element 7 (Loss rate, Float): Packet loss rate
Element 8 (Frames lost, Float): Number of frames lost
GetVideoSenderStat()
This returns the video sender statistics of the currently selected line (see SipCommCtl.SelectedLine) as a
VB array of eight elements.
Element Description
Element 1 (Codec, String): Video codec
Element 2 (Bit rate, Float): Video sending bit bit rate
Element 3 (Frame rate, Float): Video sending bit frame rate
Element 4 ((Frames sent, Integer): Number of frames sent
Element 5 (Frames dropped, Integer): Number of frames dropped
Element 6 (Packets sent, Integer): Number of packets sent
Element 7 (Frames lost, Float): Number of packets not received by remote user
Element 8 (Packets rate, Float): Packet loss rate
GrabLine(nLineToGrab)
This reserves a line to make a call. There can only be one line being reserved at a time. A reserved line
is released once ReleaseLine is called. If there is an error reserving the line, -1 is returned; otherwise, the
line being reserved is returned as an integer.
nLineToGrab:
Specifies a line to be reserved. If this is –1, the control will choose a line that is not in use and return that
line as Integer.
iOS:
GrabLine(nLineToGrab, *nLineGrabbed)

OpenSession(nLine, chAudioPayload,
chVideoPayload, nAudioRTPChannel,
nAudioRTCPChannel, nVideoRTPChannel,
nVideoRTCPChannel, bool bCaller)
Open audio video in a session.
nLine:
The line
chAudioPayload:
Audio payload type in character format
chVideoPayload:
Video payload type in character format
nAudioRTPChannel:
Audio RTP channel
nAudioRTCPChannel:
Audio RTCP channel
nVideoRTPChannel:
Video RTP channel
nVideoRTCPChannel:
Video RTCP channel
bCaller:
true if caller, false if callee
chVideoPayload should be set to 127 if there isn’t to be video.

ReleaseLine(nLine)
This un-reserves the line nLine which was grabbed by the method GrabLine.
nLine:
The line to be released.
ResetAudioReceiverStat()
This resets the audio receiver statistics.
ResetAudioSenderStat()
This resets the audio sender statistics.
ResetVideoReceiverStat()
This resets the video receiver statistics.
ResetVideoSenderStat()
This resets the video sender statistics.
SendDTMF(sKey)
This sends the DTMF message corresponding to a specified key to the current call (specified by current
value of SelectedLine). If no call is established at SelectedLine, nothing is sent.

The DTMF mode can be selected using the property DTMFMode.
sKey:
A valid key specifying a DTMF to be sent
SetNATTraversalServer(nServerType, bstrAddr, nPort,
bDone)
This function initializes the underlying AnyFirewall Engine by setting the necessary servers for performing
firewall traversal.
nServerType:
An enum indicating the type of server to be set
bstrAddr:
IP address of the server
nPort:
Port of the server
bDone:
This is false if more servers remain to be set, and True if this is the final server to be set. Firewall
detection will start after all servers have been set.

Server Type Description
eServerSRV
The DNS SRV domain name that is used to locate the STUN, TURN and
STUN-Relay/TURN servers. The port parameter for the server type is
ignored.
The following DNS SRV queries will be made:
_stun._udp.<srvdomain>STUN server (UDP)
_stun._tcp.<srvdomain>STUN server (TCP)
_turn._udp.<srvdomain>STUN-Relay/TURN server (UDP)
_turn._tc alive p.<srvdomain>STUN-Relay/TURN server (TCP)
eServerHttpProxy The HTTP Proxy server, for users using a proxy server
eServerStunUdp* The UDP STUN server
eServerStunTcp* The TCP STUN server
eServerStunRelayUdp* The UDP STUN-Relay/TURN server
eServerStunRelayTcp* The TCP STUN-Relay/TURN server
*These parameters will be used if eServerSRV is not set or the DNS SRV lookup fails.
SetSipEventHandler(sipEventHandler)
This sets the event handler for handing events from the SipComm control. This must implement the
SipEventHandler interface. The SipEventHandler is an object which implements SipEventHandler. Please
check the sample application code.
SetTURNUsernamePassword (sUsername,
sPassword)
This sets the authentication information for the TURN server.
sUsername:
Username of the TURN server

sPassword:
Password of the TURN server
This function should be called before bDone is set to true in SetNATTraversalServer.
SipCommInit
This initializes the SipComm control.

3.6. ABE: Notification Events
Notification Events
OnFirewallStatusChange(aFirewallInfo)
This fires when the firewall status changes.
aFirewallInfo is a VB-array containing following elements:
Element Description
Element 1 (Line, Integer): Line associated with the call
Element 2 (Display Name,
String):
Displays the name of the remote user in the call
Element 3 (Target URI, String): URI of the remote user in the call
Element 4 (Video call flag,
Boolean):
True if it is a video/audio call or false if it is an audio only call

Element 5 (Status, String):
1. Firewall status is shown as one of the following strings:
"Firewall type: Unknown",
"Firewall type: None",
"Firewall type: NAT",
"Firewall type: TCP only",
"Firewall type: Proxy" or
"Firewall type: Blocked" when SDK detects the firewall type.
The Line will be -1 at that time.
2. The following strings will show the status of a call:
“Trying”: Trying to traverse the firewall.
“Success”: Firewall traversal succeeded
“Fail”: Firewall traversal failed
“ICE check successfully completed”: Firewall traversal succeeded using
ICE.
“Peer-to-peer”: Firewall traversal succeeded and call completed peer-
to-peer.
“UDP Relay”: Firewall traversal succeeded and call completed using
UDP relay.
“TCP Relay”: Firewall traversal succeeded and call completed using
TCP relay.
“HTTP Relay”: Firewall traversal succeeded and call completed using
HTTP relay.
“Relay error”: Firewall traversal failed to use relay.
OnVideoSizeChange(aSizeChangeInfo)

This is fired when ImageSize property is set to change the image size to be captured and when capture
video size is changed on the remote end.
aSizeChangeInfo is a VB-array containing the following elements:
Element Description
Element 1 (Window handle,
Integer):
Window handle of the video container that has size changed (N/A on
iOS)
Element 2 (Width, Integer): Width of the video after size change
Element 3 (Height, Integer): Height of the video after size change
Android and iOS: Element 1 is ignored.

3.7. ABE: Error Handling
Error Handling
Windows
SipCommCtl returns the error code 0x80004005 (in hexadecimal format) as well as error descriptions for
all errors. These errors are thrown as exceptions in JScript. The following sample code shows how to
handle these errors in JScript:
try
{
SipCommCtl.GrabLine(-1);
}
catch(e)
{
alert(“Error description:” + e.description);
}
Android
SipCommOnError(String sError) event is fired.
iOS
OnError(const string &sError) event is fired.

4. ABE: Legal and Contact Information
Legal and Contact Information
Confidential Information: This document contains confidential and proprietary information. The document
has been provided to you in your capacity as a customer or evaluator of Eyeball Networks Inc.'s
products. Unauthorized reproduction and distribution is prohibited unless specifically approved by Eyeball
Networks Inc.
Eyeball, Eyeball.com, its logos, AnyBandwidth™ and AnyFirewall™ are trademarks of Eyeball Networks
Inc. All other referenced companies and product names may or may not be trademarks of their respective
owners.
For more information visit Eyeball Networks Inc. at http://www.eyeball.com.
Department E-mail
Sales sales@eyeball.com
Technical Support techsupport@eyeball.com
Corporate Headquarters:
730 - 1201 West Pender Street
Vancouver, BC V6E 2V2
Canada
Tel. +1 604.921.5993
Fax +1 604.921.5909

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