SIP softphone user guide
Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used by voice over IP providers. Nowadays it becomes an industry standard for audio and video communications. This softphone implements SIP as specified by RFC 3261 and compatible with all VoIP providers that follows this standard.
To start using SIP phone you should signup for a SIP account with one of the VoIP providers. Softphone can also be used as an extension of IP PBX which supports SIP. List of recommended VoIP providers can be found at the end of this guide.
Configuring SIP account
To start using softphone, in most cases, it’ll be enough to fill fields on “Accounts” tab.
“Your name” – you can specify your real name or phone number that was assigned to you by a SIP provider. In some cases this string may be presented to a called party.
“User name” – usually assigned by your SIP provider and used for authentication. It can be your PBX extension number, string or DID (phone number).
“Password” – assigned by the SIP provider and used for authentication.
“Domain” – domain name or IP address of your SIP provider.
“Register” – most SIP providers require registration to be turned on. It’ll allow SIP provider to route incoming calls to your softphone.
Multiple SIP accounts
This softphone supports multiple SIP accounts. To add another account on “Accounts” tab you can use drop down list with accounts and choose “Add new account”. Account that should be used for a call can be selected on “Dialer” tab. Selected account will be used for calls initiated from “Dialer”, “Contacts”, and “History” tabs.
If softphone is configured with multiple accounts, one of them can be made “default” at “Accounts”->“Advance settings” page. Default account will be selected in drop down list on phone start.
Upper line of the softphone is a status bar were the program displays various status information. It shows account registration status, call completion errors and other information.
Button with three vertical dots is an action menu. It is available for all tabs and provides context specific actions. E. g. for active “Calls” it provides possibility to dial DTMF tones. For “Contacts” it provides possibility to dial, add, edit, and remove contact. For “History” tab it allows to find more information about call, play call recordings, redial, add number to contacts.
After you configure at least one SIP account you’ll be able to make calls. On “Dialer” tab you can specify the number to dial and press “Call” to initiate a call. Number can be entered in dial input or dialed with key buttons followed by “Call”.
Also call can be initiated from the “Contacts” tab. First choose one of your contacts and then choose “Call” from the action menu.
Another possibility to initiate call provided at “History” tab. First choose one of the calls and then choose “Call” from the action menu.
Multiple simultaneous calls
Softphone provides possibility to make multiple simultaneous calls. You can originate calls or answer inbound calls.
To originate another call, you can switch to “Dialer” tab and dial the number from there. Alternatively call can be originated from “Contacts” or “History” tabs with an action menu.
Some SIP provider may limit you account to one or two simultaneous calls. Please consult them for this feature support.
When you originate or receive a call it appears in “Calls” tab. When you have multiple calls, active call will be outlined with a blue frame. This call is connected to your microphone and speaker. When one call become active the other calls are automatically put on hold. To switch to another call and make it active you can tap the call anywhere in call area.
The following actions available for active call:
“Record” – enable call recording.
“Stop Rec” – disable call recording.
“Mute” - mute your microphone. When the call is muted, other party wont hear you and you may still hear what they say.
“Unmute” – unmute your microphone. Other party will hear you again.
“Hold” – put call on hold. This feature should be supported by your SIP provider or IP PBX. When call is on hold, the other side will usually hear the music on hold. Please consult with your SIP provider before using this feature.
“Resume” – resume the call which was previously put on hold.
“Hangup” – disconnects the call.
DTMF tones on call
DTMF tones are often used for telephone menu navigation or for sending special call control sequences. DTMF tones can be send to an active call from a “Calls” tab action menu.
You can start call recording pressing “Record” button on the established call. To stop call recording press “Stop Rec” button.
Recorded calls are marked with an icon showing “magnetic tape” in a “History” tab. You can play them using the action menu and control the playback with “Play” and “Stop” buttons.
When you delete a call from “History”, recording is deleted as well.
On Blackberry platform recordings are stored in “shared/voice/callrecording” folder. They are saved in a WAV format and can be accessed by other applications.
If you have more then two established calls you can combine them into a conference.
Tap one of the calls and start dragging it to another call. Release it when it’ll be over required call. It’ll combine these two calls into the conference and you’ll be the third participant. Other calls can be dragged to the conference in a similar way.
During the conference call your device will be mixing audio of all participants. So take into consideration your device performance and available network bandwidth.
Conference call controls
While in conference, the following conference controls are available:
“Record” – record the conference call. Recordings will be attached to each call participating in the conference.
“Split” – split the conference. After pressing this button all calls will be removed from the conference and your device will still hold all these calls. You’ll be able to switch between them or combine them in the conference again.
“Mute” – mute your microphone from the conference. All participants can still talk to each other. Pressing this button again will unmute your microphone and you’ll be heard in the conference.
“Hangup” – hangup all calls in the conference.
When the call is in the conference the following call controls are available:
“Mute” – mute a call from the conference. Audio from this participant wont be heard in the conference, but he’ll be able to hear the conference conversation. Using this button you’ll be able to provide “listen only” experience for participants.
“Kick” – removes a call from the conference. This call remains established and you’ll be able to switch to this call with a tap in a call area.
“Hangup” – hangup a call. Call will be removed from the conference and hangup.
Basic operation on Contacts are available via action menu.
On Playbook platform version 2.x, softphone keeps its own database of contacts in application data directory.
On BB10 platform softphone integrates with the device contacts. It allows to access contacts created by another applications.
This screen displays made and received calls. It allows you to browse calls, find more information about calls, access call recordings, and call back.
Icons with arrow pointing up represents outbound calls. Icons with arrow pointing down represent inbound calls.
Icons with red “x” represent unanswered calls or calls with errors. Icons with “magnetic tape” represents calls with recording.
Various actions available via the action menu. It allows you to call number from call history, play recording, add new contact based on a call, remove one or all history records. When you remove a call from “History”, recording is removed as well.
Advanced account settings
This screen allows to fine tune account settings. It provides possibility to configure:
– authentication credentials which may be different from the username.
– SIP registrar
– outbound SIP proxy
– NAT traversal
Using “Account”->”Advanced settings” screen, account can be “Enabled” or “Disabled”. Enabled accounts are active and can be used for dialing and for inbound calls. Disabled accounts are not active and only available at “Accounts” screen for further configuration.
Using “Account”->”Advanced settings” screen, account can be made “Default” account. This will make account selected in drop down list at “Dialer” tab on softphone start. Only one account should be set as default.
Help and General softphone settings are available in the application menu. To access this menu, swipe down from the top frame onto the screen.
On this screen you can find web links to the softphone support resources. This screen can also be used to troubleshoot the softphone behavior, connection problems and provide technical information about softphone operations. In addition to high level logging it is possible to enable logging of SIP messages clicking on “SIP messages” check box. Information from this screen can be extracted using “Copy” button and pasted to another application.
This screen provides access to granular configuration of the softphone. It gives you possibility to control port to which softphone binds, specify transports: TCP, UDP. It also allows you to configure used media codecs, and codecs priorities, and aspects of NAT traversal using STUN.
In most cases these settings should be left in their defaults.
Most VoIP providers support G711u and G711a audio codecs. These two codecs are the safest choice when you are configuring a new SIP account. If you are making calls over mobile connection Speex-nb codec is a good alternative with a good quality and low bandwidth requirements. iLBC codec can be used on a low quality connections with a high percentage of dropped packets. You may consult your SIP provider about supported audio codecs.
In most cases default settings will be enough to allow softphone to work behind the NAT.
More granular control for NAT traversal is available at “Account”->”Advanced settings”->”Transport/NAT” and “Settings”->”STUN” screens.
Application menu “Help” screen can be used to find more technical details about phone operations.
Softphone web site: http://taki.sourceforge.net/ provides information on application updates, configuration tips, this documentation and other helpful resources. You can also mail your question to arkadiam AT users.sourceforge.net.
For Blackberry developers. Invoking Softphone from other applications
On the Blackberry 10 platform Softphone can be invoked from the other applications via invocation framework. Softphone target description listed bellow:
<invoke-target-name>Taki SIP phone</invoke-target-name>
<property var=”uris” value=”sip:,tel:”/>
SIP providers list
Provides VoIP service since 2004. Allows to make long distance calls on competitive rates, voice mail and conference service. Phone numbers from more then 60 countries, call back and SMS services. With every signup or recharge you contribute to open source projects.
– SIP2SIP http://sip2sip.info
SIP2SIP provides free server support for SIP signaling, RTP media (Audio and Video), NAT Traversal (including ICE), MSRP media (IM and File Transfers), Presence and XCAP (SIMPLE) and Multiparty Conferencing with wide band Audio, IM, File Transfers and Screen Sharing. This service runs on SIP Thor platform. The SIP accounts is reachable from any open network and you establish session to SIP and XMPP addresses or phone numbers. You can use multiple devices located on the public Internet or behind NAT routers.
– IPTEL http://www.iptel.org/service
Many people use IPTEL services for software/hardware interoperability testing or just as a way to call other people. The service allows incoming and outgoing calls from/to any other IP Telephony services (note: some commercial services stop calls to other Internet-based services).
Provides free SIP address, possibility to call other Ekiga users for free, public and private conference rooms.
Copyright © 2012-2013. Dmytro Mishchenko