Documentation

SIP softphone user guide

About SIP

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.

 

SIP account

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.

 

Status bar

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.

 

Action menu

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.

 

Making calls

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.

 

Active call

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.

 

Call controls

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.

 

Call recording

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.

 

Conference calls

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.

 

Contacts

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.

 

History

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

 

Enabled account

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.

 

Default account

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.

 

Application menu

Help and General softphone settings are available in the application menu. To access this menu, swipe down from the top frame onto the screen.

 

Help

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.

 

Settings

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.

 

Voice codecs

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.

 

NAT traversal

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.

 

Troubleshooting

Application menu “Help” screen can be used to find more technical details about phone operations.

 

Support

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 id=”com.utils.taki.uri.call”>

<invoke-target-type>application</invoke-target-type>

<invoke-target-name>Taki SIP phone</invoke-target-name>

<icon><image>res/icon.png</image></icon>

<filter>

<action>bb.action.OPEN</action>

<mime-type>application/vnd.taki.phone.startcall</mime-type>

<property var=”uris” value=”sip:,tel:”/>

</filter>

</invoke-target>

 

SIP providers list

Commercial:

- Diamondcard.us
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.

Free:

– 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).

Ekiga.net
Provides free SIP address, possibility to call other Ekiga users for free, public and private conference rooms.

 

Copyright © 2012-2013. Dmytro Mishchenko

 

82 Comments

  1. Just like to say thank you very much for creating this app. I was wondering in the updates that you may have a plugin for dialing from calendar, contacts or browser like the skype add on for firefox. I wrote a review as well giving you 5 *s on bb app world.

    • Integration with Phone address book is coming for BB10 platform. As for PlayBook devices it is not exposed for developers, so not too much can be done here.
      Thank you for your feedback!

  2. Hi! I have a Blackberry PlayBook. Purchased this app. How do you make a phone call with this? I know nothing about SIP phone.

    • Hi Paul, you need to open account with one of SIP providers and configure Taki to use it.
      Depending from where you want to call here is list to choose from:
      http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential

      I recommend to use one of these: Service providers.

      • how do I configure Taki with the information that I received from ekiga?

        • 1. Please register at Ekiga.net.
          2. Username: your_ekiga_username
          Password: your_ekiga_password
          Domain: ekiga.net
          Register: [checked]
          [Save]
          3. General Settings
          3.1. Media
          disable everything except
          seex-nb and g711 u-law codecs.
          3.2. STUN
          Check use STUN
          use: stun.ekiga.net in input box.

          • Hi, I am still trying to get ekiga to work on playbook. Error response: registration error. /what am I missing from your instructions? Is anyone else having this problem? Kindly send advise.

          • Please make sure you have configured a STUN server as explained at this page. Let me know via email if you still not able to make it working.

  3. As a network/nortel/asterisk administrator both your ssh and sip client make testing and remote administration so much easier!

    Thanks

    With the sip client is there a way to do +gain rx/tx, playbook is a bit quiet. I can up the gain in the asterisk configuration for the playbook but on some systems I don’t have access and would like to do it in the client.

    Thanks again.

    • I need to look at volume controls. It’s in my todo list for the future releases.
      Enjoy the apps. Thanks!

  4. I have tried this app with my Ozeki Phone System XE (http://www.ozekiphone.com),
    and it works wonderful!
    However, I also have the same issue on my Blackberry, could you manage to
    solve the problem?
    Thanks,
    Chloe

    • Hello Chloe,
      Are you talking about your existing Blackberry phone?
      Taki is available for BB10 and Playbook platforms and doesn’t support old Blackberry 7 systems.

  5. I downloaded and paid for Taking sip client for the BlackBerry Playbook and configured it to work with voipbuster and cannot get it to work. I can occasionally get it to dial a number and work but it will not do that consistently.

    It is too bad because I do like the user interface.

    By the way I have tried modifying most of the settings with no better results. Downloaded from BlackBerry appworld so it must be the latest.

    Downloaded Kuzaranda sip client from BlackBerry appworld and it worked right away.

    Good thing it was only $1.95

    If I am missing something or you have any suggestions I would appreciate an email.

    Regards,

    S. Ferrari

  6. Taki 1.3.0.13

    I have downloaded this today from the appstore and now have a problem i didnt have with 1.2. Anytime i make receive calls i get no ringing audio, no input audio and no output audio. I made a test call to my cell and got no audio on either side.

    I use voip.ms.

    • Rebooted and now everything is fine again.

      • Hi there!

        I’m having the exact same issue.
        I’ve been staring at hundreds of call logs, everything looks ok etc. Doesn’t work.
        But after a reboot everything’s back to normal.

        I figure this should be an OS related issue, perhaps after an BB World install/remove? Or could it be that Taki somehow forgets/is denied to open the RTP ports for the calls?

        Developers of Taki, feel free to contact me regarding this. :)

        Thanks!

        • I should perhaps mention that I am using the phone in a BES environment, directly connected to the SIP-server over wi-fi.
          It could have impact on the routing.

        • Thanks for reporting this issue.
          It seems to be some sound issue in BB API or underlying lib used by Taki. For now other customers confirmed that reboot solves this issue. I’m looking more attentively at this problem now and will post more details.

  7. Release notes?

  8. Hi,
    I have downloaded the Playbook app, and have registered for SIP account with
    getonsip.com
    I have configured the account info, and when I hit save, the app indicates that the account is registered and is READY.
    When I try to make a call I get an instant Call Ended Not Found message…what am I missing?

    • It means getonsip expects you to dial the number in some other format. E.g. for USA numbers they may expect it to be 13031231234 instead of 3031231234 or vice versa.

      • I am getting the call not found error as well even if I put a 1 in front of the number.

        • Hello Jeff,
          please mail me me directly arkadiamail AT gmail com details of your issue and we’ll resolve it.

  9. Hi,

    There is any possibility to use G729 codec?

  10. Hi, I purchased and I am trying to configure for Skype. I put my Skype name, correct password and Skype com. I am getting error: “Registration Error”. Can you please help? Thanks

  11. Where is the outgoing volume setting? BB Playbook is so sensitive that called party hears much echo.

    Thanks

    • There are no separate settings for volume/mic. Please use PlayBook native volume controls to adjust it.
      You’ll have much better experience if you’ll be using headphones.

  12. Hi, I purchased BlackBerry Z10 and configured Taki for making Voip calls. I have registered SIP2SIP software for SIP as well. When I dial a number out, dialler waits and timeouts, giving registration error. Please guide me to fix this problem.

  13. Will this SIP client work over a BES connection for behind the firewall access to our office iPBX?

    • I can’t definitely answer this question. Theoretically it should, if your BES server provides TCP/UDP based communication. I have no access to BES environment and no way of testing it and recommending working settings.

  14. Just purchased this and got it configured for a co-workers Z10. Works very well and I’m very impressed. A couple of questions about the contact integration.

    1. It’s great how the phone contacts are visible in the Taki interface, but for someone with a lot of contacts (my co-worker has many thousand of contacts on his phone) it can take a long time to scroll to the right contact. Any plans to add a search function, or at least a jump to letter functionality?

    2. Is it possible to launch the SIP dialer from another application (like the native contact app, or from a phone number on a web page)?

    • - I’m planning to add search function for contacts tab.
      – It’s possible to call Taki from other application using invocation framework and it is described in Documentation. In current state this possibility is more suited for application developers then end customers. I’m looking into other possibilities to integrate it closer with other BB apps including Contacts.

  15. Thanks for the information. Glad to hear that both items might be possible in a future version. One last question. I’m unfamiliar with how BB10 handles multitasking vs. background processes. Does Taki need to be an active window in order to receive calls, or is it possible for this to run as a process in the background?

    • It’s running in the background too. You can “minimize” it while on active call and switch to other tasks. It’ll also keeps waiting for inbound calls while you are in other apps.

      • Thanks again for the reply. Does the app have to stay as one of the 8 “Active Frames” that run even while minimized? Or can I close the Taki Active Frame and still have it listen for calls?

        My understanding from doing a bit of reading is that Blackberry hasn’t opened up BB10 to allow third parties to run in the background as a process just yet, although that might be coming in the future. But I haven’t been able to actively test on my co-workers phone yet.

  16. Does this work over Cellular? I am having issues getting this to work on a BES10 connected phone. Are there issues with this?
    Using simonics.com for a GV gateway.
    shows ready but when I dial out it “times out” and then show “registration error” next to account.

    • Also when I turn off my WIFI it dies and when I launch it fails and quits.

      • There will be a fix for this issue in new release. Thanks for reporting it.

    • It works over cellular. Did you make it working over wifi with your provider? I would try it first as it’s easier to troubleshoot if something goes wrong.

  17. Sir,

    I have a Net2Phone SIP account, and have set it up. Whenever I try to dial, it tells me that the number cannot be used (maybe it is the dialing format – I used +xx yyyyy, where “xx” is the country code).

    Other than this, some other comment on the integration with BB10’s contact list – most people have more than one phone number, but Taki only displays one number from the Contacts.

    • I’m not sure what Net2Phone service you are using. According to this page:
      http://web.net2phone.com/business/siptrunking/siptrunking_faq.asp

      That’s the way how they want you to dial:

      Which dialing plan should I use when placing calls?
      You should use NANP format ( 1+10 digit on US calls and 011 + CC+ on international calls).

      As for contacts, the latest version should give you access to all numbers of the contact. If this is not a case, please make sure you are running Taki ver. 1.6, take a screenshots of a contact with one number only, as you see it in Taki and in native Blackberry Contacts application and mail it to me. Thanks!

  18. Hello, I have a Voipbuster account which I generally use to send text messages. I rarely use it for calls. Unfortunately, there is no app (yet?) for the BB Q10. Will the Taki app allow me to send text messages through my Voipbuster account?

    • Taki is available in app store for Q10 devices but sending message is not supported yet. I have this feature in my plans.

  19. Hello, I’ve just purchased Taki for use with my voipfone.net account in the UK, got it registered ok. There seems to be an issue with calling many kinds of telephone number, eg ‘155’ to test the service – numbers when dialled in the app are presented in an unusual format for Uk, followed by “Call ended” and “Not acceptable here”. I was able to call a mobile.
    Any clues as to what is going on? Is this a US-only app?

    • Hello Martin,
      “Not acceptable here” – is the error messages returned by your VoIP provider. Please consult with them if 155 is a valid number.
      As for formatting numbers, Taki formats numbers as (xxx)-xxx-xxxx. Please let me know what if the common format in your country, I’m looking into improving number formatting in the next version.

      • Hi, thanks for the clues. 155 is definitely valid, it’s their service test number, I usually use linphone on the desktop and for example you can just type the SIP address, eg sip:155@sip.voipfone.net.
        I turned on SIP logging to see if I could understand it – I can see the correct sip: addresses being called, then “INVITE” and later “CANCEL” messages. In the middle, there is the pjsua code failing with errors PJMEDIA_EINVALIDOPT then PJMEDIA_SDPNEG_ENOMIEDIA.
        Is the above consistent with the sip server refusing the call (when there are correct sip: addresses used?)
        Hope this helps. I can send a section of the log if further investigation is useful.
        Re UK formatting, see http://www.area-codes.org.uk/formatting.php. There is even a section for programmers!
        Thanks.

        • Thank you for the link. I’ve already changed the way how numbers formatted to behave in the same way as they are formatted on iPhone, Nokia, Z10 phones. Any of these phones don’t format numbers that start with ‘0’.
          You can send me a SIP trace to arkadiamail AT gmail com.

  20. Hello, I’ve just purchased Taki for use with a server SIP inside my office. All work fine but i haven’t found a transfer command when i have multiple call. Will the Taki app allow me to transfer call in the future ?

    Thanks.

    • For now there is no dedicated button for transfers. There are plans to add it in the future, possibly configurable for customers that need this feature.
      Right now there is a high chance that you can do transfers with DTMF sequences. E.g. some PBX allows you to transfer calls with star codes, e.g.: *7extension. Please check your PBX documentation for this possibility.
      To send DTMFs while you are on call you’ll need to use an action menu (button with three dots).

  21. Hello,
    I downloaded the source of taki from http://sourceforge.net/projects/taki/files/ and
    I saw that the latest updates are older than 7 months. Will be updated?
    Also is there any document that shows how to use these sources with Cascade and especially with the latest version of PJSIP 2.1 that has official support for BB?

    Thanks.

  22. Hi,

    Great App, work very smooth over VPN with company PBX.

    Are there any plans for video integration?

    Regards

  23. Hello

    I have purchased this app for my z10. Using Callcentric SIP. It registers fine and I can dial normal numbers in the USA. If I dial a 1877 or 1800 number it returns “that is not allowed here”.

    This is not an issue with my SIP provider, I can use Acrobits on my iPhone and mark any 1800 or 1877 with the same SIP provider no problem. CallCentric says it’s an issue with the application I am using.

    Why does this happen?

    • If you pull menu from the top of the screen, you’ll see Support button. Please open Support window and send me the log and we’ll see what is the issue. arkadiamail AT gmail com

  24. Hello…i have purchased Taki in My BlackBerry q5…i am having voipdiscount.com accouny in qatar…when i am trying to register with this account its showing registration error…and i cant make any calls..please advice Wat to do….

    • Please double check your username, password and domain name.
      If you pull from the top of the screen, there is a Support window. You can send me output from this window and I’ll be able to advise. arkadiamail at gmail dot com.

      • Just purchased TAKI and it does not work on an q10 with a VOIPdiscount SIP account

        How can I get a refund

  25. Does taki support caller id on blackberry z10

    • Yes it does. CallerId is your SIP provider related feature, as they have full control over it. In Taki you can specify desired Caller Name and Number.

  26. Hello, will this work over LTE?

  27. one last question; will the phone accept incoming calls when the screen is off? I don’t imagine so but I can hope…

  28. When dialing a number Taki will dial out but as soon as speaker is selected you lose all audio. Switching back to the phone audio makes no difference. Call is done by then and nothing can be done to retrieve it once you hit speaker.

    • What version of OS are you running?
      Taki 1.9.4 has a fix that should fix one audio issue for 10.2.1.

  29. Hi, I love the app! I am using it with my flynumber.com phone numbers. The only thing I haven’t figured out is how to make the phone vibrate instead or rings. When I am in meetings, I can<'t leave the ring on, so I need to turn down the volume, but then I don't know if someone is calling. Is there any way to make taki vibrate when a call is coming in?
    thanks,
    Kevin

    • Hello Kevin,
      in current version there is no such possibility. I’ve added your request to my todo and will try to address it as time permits.
      Thanks for your suggestion.

  30. Hi. I have been unable to configure my Skype account to use it with Taking. Is Skype compatible with Taking? I have a paid subscription with Skype to call unlimited to Mexico. Taking keeps telling there’s a registration error.

  31. Hi,

    Is it possible to change the Local or Listen SIP/RTP ports used by Taki?
    Thanks,

    James

    • It’s possible to set alternative port for SIP. Settings->Transport->Port

  32. Hello together,

    while recently searching for a SIP app that fulfils my needs on my Q10, I fell across Taki. However, before buying I have on simple question:
    Does Taki integrate its call log into the BB call log and hub? Since I very much use the BB10 hub it is vital for me that all calling information etc. is consistant and available in one place. So are the calles made and received through Taki logged in the BB internal call log ? (same for sms sent through Taki)

    Cheers,
    Ludwig

  33. Hey there!
    Nice app…I only prefer using native apps so I immediately bought yours when I saw it.
    I’m in Germany and use 1und1 as my provider. They provide credentials which I’ve input and everytime I launch the app I see “initializing” and then “ready” and then an immediate crash. I’ve deleted and reinstalled so I can start over but it’s the same thing everytime. I don’t even have time to switch to the log to see what’s happening. Where are the logs stored so I can take a look at them?

    KH

    • Please turn an Airplaine Mode and start Taki. Then if you pull from the top of the screen you should be able to open Support with application logs.
      It’ll also allow to reconfigure account if required.