RTCOMM

Basically I cannot get my RTCOMM SIP to authenticate with Gizmo's servers (using an account I have had and used since 8/2007) with the UDP transport selected in RTCOMM. I must use Auto or TCP in order to achieve authentication (login).

Argh- I understand this is beta, and I worked on this for two days. I tried all kinds of things, and failed to succeed in getting RTCOMM working 100% (but I did get it partially working soon after I succeeded with authentication).

I think I have read almost everything there is to read on RTCOMM, SIP and GIZMO these past two days. I have been to every forum (incl. this one) searched for anything related to SIP and the N800, including GIZMO. I have spent hours reading FAQ's and googling everything I could on RTCOMM, SIP, GIZMO, GOOGLETALK, XMPP, JABBER....you name it. I read all about the difference between UDP and TCP. I even spent long periods of time reading IRC Logs where our guru ZeroJay talk to other folks about RTCOMM on the maemo platform. I ran software which tested my WRT54G to see if the ports were open and if my WRT54G was symmetrical NAT or assymetrical NAT (that test showed my router used symmetrical NAT). I checked the http://www.iana.org/assignments/port-numbers website to verify 5060 and 3478 ports are correct.

ZeroJay here says RTCOMM works great, so I didn't give up at trying to make it work for me. I have spent alot of time on this.

I have a simple network at home. My broadband terminal adapter (Time Warner's Road Runner service) is connected to my Linksys WRT54G Ver4 with std firmware (there are no other switches,hubs or routers). That setup has worked flawlessly for several years. I am very familiar with the configuration setting in my Linksys, and doing things like port forwarding, enabling Upnp, DHCP, MAC filtering, WEP/WPA (when things need trouble shooting I shut off all encryption until I'm done), bla bla bla.

Here is what the problem is:
#1- Nokia and other say to put my SIP number and password into RTCOMM, and the correct SIP Proxy and STUN server (and ports), and chose UDP as the transport ...I did this and RTCOMM refused to authenticate with the SIP server (Gizmo's). Thru much trial and error I determined that RTCOMM would NEVER authenticate with my SIP number.... the server required my real username (not SIP number).
#2-not only did I need my username and password (not SIP number) to authenticate, but RTCOMM also refused to authenticate with UDP selected.....either I had to chose "Auto", or TCP, or TCP via HTTP proxy.
~~~~~~~~
The authentication problems solved I can call my cellphone or other PSTN phones and they hear me, but I hear nothing. I tried all kinds of setting changed, router port forwardings, and so forth. Thinking there was some sort of conflict maybe with Googletalk, or GizmoProject I made sure those weren't running on any computers in my network, and I made sure they were no running on my N800. I must have made 50-80 call over the past two days of debugging, carefully changing one thing at a time.

I also tried using RTCOMM to call other SIP clients with the same result-- it rings there, they answer, they here me, I don't here anything. It as if the audio out on my N800 is muted, but only in (during) the SIP call using RTCOMM. In fact, I hear a couple ring sounds on my N800 when I place the call, then the ringing sound stops (before the other party answers)....so that may be a clue as to what is happening.

Am I missing a critical codec? Where would I look?

BTW- Gizmo 3.1.0.78 works on my N800. It is RTCOMM that only half works. I know it is beta and all, however ZeroJay proclaims it works fine, so I became obsessed with getting it running.

I have never flashed beyond IT-OS2007 (4.2007.38-2) so RTCOMM 1.3-3 is supposed to work, right? What is up with the misinformation about the authentication problems I had early on? Why can't I autenticate with RTCOMM set to UDP transport? WHy must it be TCP or Auto? Is that a clue as to why they I can't hear them? Doesn't seem so to me because they hear me. Is my N800 routing the audio away from the speakers to something else? (bluetooth? ...nooooo). It acts like it is. This problem seems like a simple audio (speaker) problem, like RTCOMM is getting the packets, making the sounds but sending the sounds somewhere other than my N800 speakers ( I tried the wired headphones too, no solution). My N800 works properly in every regard (except RTCOMM Beta).

I use:

proxy01.sipphone.com
---advanced---
Transport: <Auto> (or TCP)... UDP refuses to authenticate
<leave Outbound Proxy blank, Gizmo Project's equipment demand this>
Port: <5060>
Discover public address <check>
Keep alive <Auto>
Keep Alive freq <0>
STUN Server: stun01.sipphone.com
Port: <3478>


I'm stuck. I have nothing else to try. Thanks for your assistance.

Dunno, the steps on the rtcomm page worked fine for me (minus one problem with my network setup that was solved at the router). Maybe just wait for OS2008? rtcomm is now built-in and much more reliable.

Thank you General. Would you kindly tell me if you used UDP as the transport?

Also, if you happened to use Gizmo SIP server(s) with RTCOMM, did RTCOMM authenticate with your 10 digit SIP number? or did you have to use a "name". ( please disregard if you used another SIP service ).

Many thanks.

Hi, don't want to confuse you more but wasn't there some people complaining about a conflict between skype and rtcomm? (but maybe it was solved on the last updates).

I use rtcomm daily and to set it up with voipbuster and voipdiscount it took me something like 3 minutes. The reception is acceptable most of the time (there are days that it refuses to transmit the complete words though). It also annoys me that I can't check who is online.

One more thing; Did you try to also set the auto discover for the STUN server?

Chris

Thanks Chris, I tried both stun01.sipphone.com on port 3478 with that port forwarded on my router (they hear me I don't hear them), and I tried Auto-detect STUN: <check> (that grays out the server and port setting. The result was the same, I authenticate, I call a mobile phone, I hear a couple rings, their phone rings, they answer, they hear my voice, I hear nothing on my N800 speakers. I'm hoping ZeroJay will advise me as to why this isn't working right. He seems very knowledgeable when it comes to RTCOMM.

BTW- with regard to any Skype interference with RTCOMM, I was certain to not have any computer running Skype on my network, and that includes my N800, during my troubleshooting.

It is very mysterious that RTCOMM refuses to authenticate when I select UDP, as the transport because the RTCOMM site at maemo say to use that, they even use the Gizmo servers in their examples just like me. Why doesn't UDP work? Anyone have an answer? Also, why did RTCOMM not authenticate when I used my SIP number as User name: ? Maemo's examples show that also. I know it is beta but these two things strike me as very bizarre.....and I feel they have nothing to do with my biggest problem of not hearing any voice at my end of the voice calls.

I'm still running 4.2007.26-8, but I do have RTCOMM 1.3-3, and from what I understand, the last OS2007 update only patched the memory card drivers. I'd like to try to help.

I have my Gizmo account configured in RTCOMM, and it seems to work, though I haven't found a use for it. My configuration appeaars to be the same as yours, except I do use my Gizmo SIP number, not my Gizmo username, as my user name, and I use UDP (Auto doesn't work). I've not opened or forwarded ports (or anything like that) on my access point or router.

I have, however, used GrandCentral to make my test calls. (I don't know anyone who uses SIP.) Would that make a difference?

Here's what works for me: I click on my work/home phone number in GrandCentral Mobile, and it calls my Gizmo SIP number. RTCOMM opens, and I answer the call. I hear it ringing my landline, I hear my voice mail/answering machine greeting, and I can leave myself a message.

It's just basic testing, but everything seems to work. Is there anything you'd like me to try?

Taken from account setup in ITOS2008 (should be the same in RTCOMM) here are my settings.

Username: [gizmo username]@proxy01.sipphone.com
Password: [gizmo password]
Use for PSTN calls: Yes

Advanced Settings: Connection
Transport: UDP
Outgoing Proxy: proxy01.sipphone.com
Port: 5060
Discover public address? Yes
Keep-alive mechanism: Auto
Keep-alive frequency: 0

Advanced: Authentication
Authentication username: [blank]
Password: [blank]

Advanced: STUN
Auto-detect STUN? Yes

Works fine for me with port 5060 forwarded, though I have been having a problem lately with calls being very choppy that I haven't complained about yet.

(Incidentally, this has to be the first time I've been called a guru.)

Thanks guru. I hoped you'd throw your hat in the ring.

Well, I'm stumped. It is really strange that I cannot authenticate using the UDP transport. Could it be some other application interfering? I have googltalk disabled, and skype not running. Both you guys above say you use UDP (which is what maemo tips advised). I dunno. I'll keep watching this thread.

I'll get a VoipBuster account later today, and try setting their servers up with RTCOMM and test that too. Has anyone used their N800, with RTCOMM and VoipBuster (sip.voipbuster.com and stun.voipbuster.com)?

I am talking my sister into getting a N800, if my RTCOMM still is goofed up, I'll put RTCOMM on her N800 (when she gets one) before she loads all kinds of apps on it. I'll see what that shows.

BTW- I updated python runtime to 2.5 (I had 2.4) and that didn't fix anything.

Is there a previous version of RTCOMM I can try? ( one before 1.3-3 )?

Thanks to all who contributed.

ZeroJay.... I don't know how you got your's working, especially due to your statement that you put server information into the "Outgoing Proxy" section. Here is a link to a Gizmo page on configuring hardware that spells it out.

http://support.gizmoproject.com/inde...kbarticleid=83

It makes it clear on that page that if you put anything in that Outbound Proxy field, your device won't be able to connect (to the Gizmo servers).

UPDATE:
After exhausting every possible cause of RTCOMM malfunctions, I have decided to give up. Too many things don't make sense as to why this software refuses to authenticate the way Nokia say it should to sipphone.com's servers, and when I do (with considerable effort) manage to get RTCOMM to authenticate, and place a call to a PSTN number, I get one way audio. I also cannot get RTCOMM to authenticate whatsoever to voipbuster.com's servers either. I have tried everything. I am very confident now that there is nothing wrong with my LAN and router here. Everything points back to RTCOMM. Either there is a conflicting application, or missing codec on my Nokia N800 that prevents my N800 from bidirectional audio with RTCOMM.

As I see it, at this stage I have two choices, and that is to wipe my N800 clean and reflash the firmware and reinstall/retry RTCOMM, --or-- say the heck with it for now and just stick with Gizmo and Skype. I hope no one else wastes as much time as I did on RTCOMM. My recommendation to the community is this : if you can't RTCOMM working for you in the first 3 minutes, delete it and spare yourself the agony of spending hour after hour trying to get it working. Good luck to all.