Tuesday, September 1, 2009

How to use your SIP Server with Nimbuzz Client

You can use Nimbuzz Client (Mobile or PC), which is a XMPP Jingle Client to connect to your SIP Provider or Personal SIP Server(Company Asterisk or iptel.org Services for instance). This proves how powerful and inter-operable Jingle is.

There are some requirements in order to use it:
* Your SIP Server/Provider MUST be connected to the Internet in a public IP.
* Your SIP Server/Provider MUST provide or act like a RTP Proxy. (Asterisk does that by default)
* Your Nimbuzz Client MUST be connected to the Internet, through a connection that doesn't block UDP/RTP Traffic and also allows TCP connections to port 5222, which will be used by the XMPP Connection.

How to setup with your own SIP Server(General):
1) Setup your SIP Server using your domain on port 5060. Setting up a SIP Proxy is optional.
2) Enable RTP Proxy, in order to guarantee audio both ways, even when users are behind NAT.
3) Create the users/passwords.
4) Make sure to configure your firewall correctly enabling UDP in/out traffic to Internet on port 5060 and in all the port range used by your RTP Proxy (usually 15000 to 65000).
5) Log on Nimbuzz Client using your Nimbuzz Account, go to SIP Settings and use the credentials that you created previously:
Username: user01@yourdomain.com
Password: pass
Proxy: (Leave it blank if you didn't setup it)
6) Done, you now should be able to place and receive calls in your Nimbuzz Client using your own SIP Server. (Refer to troubleshooting section in case of issues.)

How to setup with using SIP Provider(iptel.org in the example):
1) Log on Nimbuzz Client using your Nimbuzz Account, go to SIP Settings and use your SIP credentials that you created previously:
Username: userABC@iptel.org
Password: pass
Proxy: sip.iptel.org
2) Done, you now should be able to place and receive calls in your Nimbuzz Client using your SIP Provider. (Refer to troubleshooting section in case of issues.)


* If you cannot even get registered to your SIP Server/Provider:
Double Check your credentials on Nimbuzz SIP Settings and also in your SIP Server Configuration.

* If you cannot get Calls completed:
Debug the SIP Signalling and also refer to your SIP Server Call Routing Configurations. (In Asterisk check for extensions.conf).
When using a SIP provider contact your provider for details of how to place calls(Number format, Valid Destinations, etc).

* If you cannot get Audio Both Ways:
Make sure your SIP Server/Provider has RTP Proxy enable. It is very likely that if you are experiencing this situation, you or the person you are calling to, is behind a NAT.
The interoperability of this service depends on RTP Proxy availability to guarantee both ways audio for users behind NAT.
Check you SIP Server Configuration or contact your SIP provider for further details and support.
Once the RTP Proxy is SIP Provider/Server responsibility.
Also make sure you are using an Internet connection that does NOT block UDP/RTP Traffic. This is a requirement. If using 3G check with your carrier about UDP/RTP restrictions.

I hope this article clarifies a little how to integrate to your SIP Services. If you already using/used another provider, please post comments about how you did it.


  1. I have a good experience with several VoIP providers; Gizmo, Voxalot, and my own server.

    For a certain range of Nokia phones you don't need to have an additional client to use SIP (as can be seen on the TruPhone website). I use an older Nokia E60 as VoIP phone over 3G (or WLAN). But Nokia had dropped SIP support for a time... for those unlucky, Nimbuzz and Fring are good alternatives!

  2. Gerard Braad, I used to use SIP embedded support on my old E61.
    But the battery use to drain so fast, I guess this also depends on Re-Registration period of your SIP provider.
    I personally switched to Nimbuzz now that I use 6120 Classic. I can stay online with presence, contact list, VoIP on GTalk, Nimbuzz and SIP for 5 hours. Feels like SIP only is not enough.

  3. Really great ideas. I like every example. Just might have to try these... So cute! Thank you!
    more templates easy to download

  4. thiagoc:

    I didn't know Nimbuzz changed their policies in using SIP accounts. Nimbuzz worked very well before, but I didn't used it because I was testing the E71 SIP client.

    As you stated in an earlier comment, the Nokia SIP client is limited in some fields. For example, if you need to talk via UMTS/HSDPA and you have some packet loss, is IMPOSSIBLE to talk. It's not a Nokia fault, it's VoIP nature.

    I prefer to use Fring/Nimbuzz because they have a (sadly, proprietary) protocol than can manage these issues. Oh yes, with delay, but at a $0 cost.

    So, as you said me in the Nimbuzz Forum, I checked all the requirements, and I already met all of them, but ONE: I configured RTP port range to 10000-11000.

    Could be the reason?

    Thanks for your posts

  5. @Nestor, I do recommend to open a wider range and most important make sure that this range is also running in a public IP and it is not blocked by firewalls. Remember to enabled NAT=Yes in your configuration files.
    Your PBX will receive the media directly from/to your phone.

    I also have one remark about your statements, Nimbuzz doesn't add any delay to Voice Streams at all. This is actually one of the reasons that makes me prefer Nimbuzz over Fring.

    Let me know which PBX are you using, so I can be more specific.

    Thanks for posting!

  6. For ActionVoip Setting on Nimbuzz:

    Password: [YourPassowrd]
    Proxy: [Leave it BLANK]


  7. I use AZZU with settings:


    Working fine for me, both way calling :)

  8. I have windows Mobile HTC HD, I downloaded Nimbuzz from nimbuzz.com - windows mobile  touch screen.
    When I am using Nimbuzz for VOIP calls, once the called number is answered, the call is terminated!!

    I am using same SIP account on my wife Nokia E71 and it is working fine.

    I think the problem in the Nimbuzz windows mobile version.

    Can any body help me to fix this problem?

  9. Hi i am Nazarov , i have a problem with Sip Provider Setting i have Username with Password & Proxy . but i have no Dmain (Provaider.net) i don't know what is meaning of domain what shold i write there ? what i shold do ? help me Please*

  10. I like the idea of being able to voice call a user on the same IM community as one which I belong to. The question or concern I would have is around gaining momentum on the platform and enticing users to use the Jabber service instead of well established P2P platforms such as Skype. white gold jewellery

  11. Hello I used this SIP provider for a long time, its really effective of course if you use correctly manner, contrary you can cause a lot of problem with your cell phone.

  12. Thank you for sharing information. It is quite useful for us also. I always love to read such type of things.

  13. Great review about this XMPP Jingle.

  14. I feel ecstatic I found you website and blogs.
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