No inbound audio with Nimbuzz and Fring

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fixup77
Posts: 93
Joined: Sun Jan 27, 2008 1:56 am

No inbound audio with Nimbuzz and Fring

Post by fixup77 » Thu Aug 30, 2012 6:41 am

There must be some NAT handling issues between SS and Nimbuzz and Fring etc. No inbound audio when NAT is involved. If I let them connect directly to the sip provider such as sipgate, i.e., phone -> nimbuzz -> sipgate, no problem. Once SS is in the middle, i.e., phone -> Nimbuzz -> SS -> Sipgate, then no inbound audio. So the problem seems on the SS side.

If the phone has a public IP (using 3g instead of WiFi), then no problem. If one phone is on 3g and the other is on WiFi, when they call with each other through SS, the WiFI one has no inbound audio.

If I don't use Nimbuzz and Fring, but use a normal SIP client like Acrobits, then no problem.

So, I think there are some NAT handling issues between SS and Nimbuzz kind services.

Aaron
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Joined: Thu Jul 12, 2007 12:13 am

Re: No inbound audio with Nimbuzz and Fring

Post by Aaron » Thu Aug 30, 2012 10:13 am

The SIPSorcery does mangle any private IP addresses it finds in incoming call requests and responses in an attempt to help NAT. Apart from that it does not take any other actions and does not proxy audio and therefore has pretty much the lightest touch possible with regards causing audio issues.

It's possible to turn off even the small adjustment the SIPSorcery makes for NAT by turning private IP address mangling off using a dial string option of ma=false.

Code: Select all

sys.Dial("1234@someprovider[ma=false]")

fixup77
Posts: 93
Joined: Sun Jan 27, 2008 1:56 am

Re: No inbound audio with Nimbuzz and Fring

Post by fixup77 » Sun Sep 02, 2012 7:54 am

Well, I set up a FreeSWITCH just to test it for nimBuzz, no NAT problem. I'm sure FS is not doing media transcoding nor proxying. I enabled "inbound-bypass-media" as described here:

http://wiki.freeswitch.org/wiki/Bypass_Media

and here:

http://devblog.brahmancreations.com/con ... sys-sipura

I know it is indeed bypassing audio, because if I disable this feature, call cannot connect at all with a g729 error (FreeSWITCH allows bypass g729 only because I don't have a g729 license).

So, there must be some NAT handling issues between nimbuzz and SipSorcery; FreeSWITCH does not have this problem under the same condition. The only difference is: FreeSWITCH is registered to CallCentric which is forwarded to SipSorcery (my free SS account allows only one SIP provider).

I tried FreeSWITCH a year ago and dropped it because I could not make G.729 codec to work with GoogleVoice dialing. Today I figured it out and now I can use FreeSWITCH to achieve the same functions of SipSorcery, i.e., seamless GV dialing and no touch on audio, G729 supported. Should be even more - FreeSwitch allows running external scripts and applications, so I should be able to make seamless CrowdCall dialing too, similar to what I did for GV dialing as described here (use an external script to initiate the call and then park the call for callback):

http://wiki.freeswitch.org/wiki/Google_Voice_API

I loved SipSorcery, but it was getting harder and harder to run it locally and the free account became too restricted and eventually over. For those who want to run something similar to SipSorcery, FreeSWITCH is an alternative, a better one. I have noticed that FreeSWITCH uses very little CPU if any at all, while SipSorcery consumes constantly 15% and more even while idle.

I hope you can solve the NAT issues between nimBuzz and SS. NimBuzz is by far the only acceptable client for smartphones, way better voice quality than Acrobits, CSipSimple and Bria etc. Yes, it does support g.729 even though only g711 and iLbc show in option (the trick is select auto).

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