cSipSimple configuration settings? *HELP*

[ryangrant]

Hi guys,

I have purchased the config settings and downloaded my SIP client of choice which is cSIPSimple because sipdroid does not support the codec g729.

I have read around the internet and seen that some users can get freephoneline.ca config file to work with cSIPSimple, however I cannot.

let me explain, I have all the server settings and account settings, the client connects, but when i make a phonecall, i cannot hear anything, the other side can hear me.

I read about STUN servers and even turning on 'ICE' mode, however both these features on did not help. I tried so many combinations over wifi and over 3g, nothing is fixing it.

one user on csipsimple site said they called freephoneline.ca and freephoneline.ca set the useragent to csipsimple.

I dont see how this would help..also when I am connected, i login to my freephoneline.ca account online and see it shows csipsimple user agent...

is there any advice? or anything you need to know? I also had just default port of 0 in my config, which seemed to still work, im guessing it just auto finds the port that its supposed to...however i searched around and found that port 5060 is the one that we should be using, there are also other ports but I only entered 5060. if its a port issue please let me know...

[bridonca]

You have a NAT issue, and sadly, Freephoneline is not very tolerant to flaky NAT's. Freephoneline need only 2 dedicated ports available to it, a sip port, like 5060 (you can use ports 5061 to 6060 if 5060 is blocked or already being used) and you need a RTP port (a high numbered port like 12000 should do because it is unlikely to be in conflict) Linux based SIP phones are still rather raw betas, and a lot are just not mature enough to handle flaky NAT devices yet.

In such a nasty NAT environment, I think you would be better off using another VOIP provider like Voxalot. Voxalot is a bit more tolerant to awful NAT's, especially the 1 way audio issues. You can use Freephonline as one of Voxalot's Voice Service Providers, you would can still make freephoneline calls. Voxalot is free, with optional paid services.

[FONGO_steve]

Please be sure to read the forum rules before posting I have moved your post out of the configuration sharing section and into a more appropriate location.

[MrDoo]

You need to enable NAT. Also try forwarding ports 10001-20000, in addition to 5060 to your device in your router. This assumes you are on wifi.

BTW, Voxalot does not support provider registration (which fpl requires) in it's fee service.

[bridonca]

Mrdoo, outgoing Freephoneline calls work just fine with my free Voxalot account. I happily pay $15 a year to get incoming Freephoneline calls though Voxalot, but I also get to use web callback, which is great if you have an awful internet connection.

[MrDoo]

Bridonca, I meant "free" service instead of "fee". Registration support is of course available to paying customers. Other free options are pbxes.org and sipsorcery.com. Sipsorcery is for advanced users, but pbxes.org is relatively simple to use. There are some usage limits to be aware of however.

[ryangrant]

well i'd prefer to figure out what's wrong.. I mean everyone talks about stun servers and ice capabilities..not sure why my carrier and my internet at home over wifi are both having so much issues with traversing NAT...

[bridonca]

What is wrong is you are not getting a direct 2 way path of SIP and RTP from freephoneline to your SIP client. When the incoming RTP audio gets from freephoneline to your router, the router has no clue on where to send that audio. That is the case because the router's current firmware is not up to the task, and that needs to be addressed, either by a firmware update, a 3rd party firmware update like DD-WRT, or router replacement. Even so, I prefer to manually assign ports to the appropriate device, because assures the correct routing path.

There are some VOIP providers, and some SIP phone vendors that employ non-standard tricks to attempt to address this VOIP through NAT problem, with mixed results. I know freephoneline.ca does not, and it looks like your soft phone doesn't either.

As for your STUN server and ICE capabilities, none of that is any good if your router is too stupid not to know where to route the traffic.

As for SIP VOIP over 3G internet, that is another can of worms. Short answer, use Skype instead. I got Voxalot working, but I was less than impressed with the results. SIP VOIP over 3G internet is hostile territory.