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2007-11-20
, 05:28
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Posts: 5,478 |
Thanked: 5,222 times |
Joined on Jan 2006
@ St. Petersburg, FL
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#12
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To be honest, every case of one-way-only audio I've ever heard of with SIP has been an issue with NAT setup. So either you've misread or misunderstood some instructions that result in the client trying to communicate on the wrong port, or your network simply wont work with the rtcomm client (and it is rather picky in this version).
Anyway, as I said, wait for 2008, it cleared up an intermittent issue I was having with call disconnects (as well as becoming generally more stable and faster), it will work much more nicely.
Anyway, as I said, wait for 2008, it cleared up an intermittent issue I was having with call disconnects (as well as becoming generally more stable and faster), it will work much more nicely.
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2007-11-20
, 13:46
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Posts: 354 |
Thanked: 93 times |
Joined on Oct 2007
@ New York
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#13
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Originally Posted by GeneralAntilles
Thank you General,
To be honest, every case of one-way-only audio I've ever heard of with SIP has been an issue with NAT setup. So either you've misread or misunderstood some instructions that result in the client trying to communicate on the wrong port, or your network simply wont work with the rtcomm client (and it is rather picky in this version).
Anyway, as I said, wait for 2008, it cleared up an intermittent issue I was having with call disconnects (as well as becoming generally more stable and faster), it will work much more nicely.
Why was it that in order to authenticate I had to use my "name" and not my SIP number? That was before I ran across the one way audio ( and yes I read alot about that one way audio and NAT issues was always given as the cause ). Why did my RTCOMM refuse to authenticate set for the UDP transport when everything (and everyone) said to choose UDP?
If ZeroJay had not said it worked well, I never would have spent so much time on it. When I had the one way audio, I spent hours trying to find what might be wrong with my small simple network. I don't really NEED it to work because I use Skype, but once I spent a good amount of time on it, I refused to quit (until yesterday).
I'll keep my fingers crossed that OS2008 (final) will be a wonderful thing but what of Skype and Maemo Mapper and so many other things which are yet to be officially compatible with OS2008. I dunno, it looks like I may be running OS2007 for many more months.
I hope everyone in the U.S. has a Happy Thanksgiving. L8r.
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2007-11-20
, 22:25
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Posts: 1,648 |
Thanked: 2,122 times |
Joined on Mar 2007
@ UNKLE's Never Never Land
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#14
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Originally Posted by xxM5xx
You didn't mention what was your settings for the voipbuster test. Don't really know how gismo works but in general for sip you need to open more ports than just the 5060.
UPDATE:
I also cannot get RTCOMM to authenticate whatsoever to voipbuster.com's servers either.
For voipbuster you need:
1. The following Destination ports need to be allowed on your firewall:
UDP 5060
UDP 11113
UDP 10300 - 10311
UDP 6901 - 6920
2. Register once using the voipbuster client.
3. put the following settings.
http://www.voipbuster.com/en/sipp.html
and those are my settings:
Username: [username]@sip.voibuster.com
Password: [password]
Advanced Settings: Connection
Transport: UDP
Outgoing Proxy: [empty]
Port: 5060
Discover public address? Yes
Keep-alive mechanism: Auto
Keep-alive frequency: 0
Advanced: Authentication
Authentication username: [username]
Password: [password]
Advanced: STUN
Auto-detect STUN? Yes
Hope it helps..
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2007-11-21
, 06:50
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Posts: 354 |
Thanked: 93 times |
Joined on Oct 2007
@ New York
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#15
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Thank you Saturn. I'll give that a try in a few days. I am also going to reflash the N800. I didn't want to because I have a ton of maps for MM on there, and a whole lot of apps. Reinstalling it is going to be such a pain. When I do it I'll do one at a time because I don't want to do a restore from backup and be in a similar situation, plus if there is one app or lib that is causing my issue, I have a good chance of identifying it using this method.
Thanks for your very detailed settings layout.
Regards
Thanks for your very detailed settings layout.
Regards
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2008-01-17
, 20:36
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Posts: 22 |
Thanked: 0 times |
Joined on Jan 2008
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#16
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Originally Posted by Saturn
You didn't mention what was your settings for the voipbuster test. Don't really know how gismo works but in general for sip you need to open more ports than just the 5060.
For voipbuster you need:
1. The following Destination ports need to be allowed on your firewall:
UDP 5060
UDP 11113
UDP 10300 - 10311
UDP 6901 - 6920
2. Register once using the voipbuster client.
3. put the following settings.
http://www.voipbuster.com/en/sipp.html
and those are my settings:
Username: [username]@sip.voibuster.com
Password: [password]
Advanced Settings: Connection
Transport: UDP
Outgoing Proxy: [empty]
Port: 5060
Discover public address? Yes
Keep-alive mechanism: Auto
Keep-alive frequency: 0
Advanced: Authentication
Authentication username: [username]
Password: [password]
Advanced: STUN
Auto-detect STUN? Yes
Hope it helps..
First post here guys...
I've done everything stated in all of SATURNS posts regarding getting Voipbuster working but I have a problem that nobody else seemed to encounter..
In the field Username: [username]@sip.voibuster.com
I put in my voipbuster user name and it doesnt accept it ..It says "field cannot contain the character"..
My user name is a 2 part name with a space between the names....i.e. {flat top @sip.voipbuster.com}
but apparently this is not allowed.The space is not being allowed by the rtcom program..I double checked my login to voipbuster on my pc using the voipbuster program and my username definitely has a space in the middle.
What do I do to get around this?
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2008-01-17
, 21:10
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Posts: 22 |
Thanked: 0 times |
Joined on Jan 2008
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#18
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Originally Posted by sondjata
Thanks for the suggestion but it doesnt work...
you can try the backslash before the space?
error:unable to establish connection
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2008-01-17
, 21:45
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Posts: 1,648 |
Thanked: 2,122 times |
Joined on Mar 2007
@ UNKLE's Never Never Land
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#19
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It's weird that someone allowed you a username with space..
I would sent them a mail to ask them to change it with an underscore. If you are still in the trial just create a new account.
I would sent them a mail to ask them to change it with an underscore. If you are still in the trial just create a new account.
| The Following User Says Thank You to Saturn For This Useful Post: | ||
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2008-01-17
, 22:27
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Posts: 22 |
Thanked: 0 times |
Joined on Jan 2008
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#20
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Originally Posted by Saturn
Its an old voipbuster account ,I've had it since 2004! .
It's weird that someone allowed you a username with space..
I would sent them a mail to ask them to change it with an underscore. If you are still in the trial just create a new account.
I have a bunch of credit left on it and alot of contacts ,all over the world,so I'd prefer not to change it..
Is it not possible to use this account with the rtcomm program?
Kicking Nokia in the jimmy, one marketing exec at a time.