Can't edit ports used

[GoogleCodeExporter]

What steps will reproduce the problem?

1. Try to edit inbound RTP ports

What is the expected output?

A means to set the set of inbound RTP ports

What do you see instead?

No UI available for this.

What version of the product are you using? On what operating system?

0.2-0.1 on Android 2.2.1 on T-Mobile 3G

Please provide any additional information below.

I'd like to use CSipSimple to connect to my Asterisk server two ways:

NATed behind a wireless router (Netgear WNDR3700) for WiFi connectivity, and 
through the public internet at a publicly routable IP address (with, 
surprisingly, no inbound NAT (I can disable that)).

I DON'T want to have to resort to STUN on the handset (it appears to really eat 
battery).

Asterisk is smart enough to distinguish between calls on the WAN IP address and 
LAN IP address (the LAN is in front of the WNDR 3700 which NATs wireless 
clients), and can ignore the addresses and ports in the SDP messages, using the 
actual remote NATed addresses and port numbers.

Now, as far as the NATed wireless clients WiFi clients on the LAN side, port 
forwarding can be used to map the RTP packets, SO LONG AS the clients use 
non-overlapping RTP ports, and probably unique SIP ports.

But, there is no way to control this in the CSipSimple configuration.

Original issue reported on code.google.com by rene.s.h...@gmail.com on 25 May 2011 at 11:39

[GoogleCodeExporter]

UI is available.

Read : ExpertSettingMode wiki page.

Keep in mind, CSipSimple is for mainstream users, I try to keep things as 
simple as possible and to show only settings used by 90% of users. If you are 
an expert, just activate the ExpertSettingMode and welcome the geek world of 
csipsimple users ;)

And you'll find that CSipSimple is the most tweakable app ;). That's just 
hidden to not afraid mainstream users :)

Also do not hesitate to read the wiki, there is a lot of hidden things you 
could learn from the app.

About RTP port, in CSipSimple (herited from the pjsip stack), what you'll setup 
is the begin of the range. By default it's 4000. You can change it in RTP port 
settings of network section. It will always use from port to port+8. (cause 
pjsip is configured to support up to 4 conversations so use 4*2 ports for RTP).
For now, no ways to split ports.

And last point about STUN and battery, you're absolutely right. Pjsip guys 
(pjsip is the sip stack that csipsimple use as main sip stack), announced that 
pjsip-2.0 (the next big step) will support - among other cool stuff- on demand 
STUN. This will incredibly save the battery cause it will not anymore ping the 
stun server while idle but only when actually necessary.

The video branch of csipsimple that early use the dev branch of pjsip-2.0 
already support that... but well not yet released cause still unstable... Was 
just to say to not worry, that's planned ;)

Original comment by r3gis...@gmail.com on 26 May 2011 at 8:12

• Changed state: Invalid

[GoogleCodeExporter]

Oh! I don't SEE an "RTP Port Settings" section in the expert Network config 
section.

Should I try a development branch version? I used the stock one. (And, I am 
anxious to try TLS).

I see the following:

Transport:
Wifi keep alive
Mobile keep alive
Resolve DNS SRV
Use compact SIP

NAT traversal:
Enable ICE
Enable STUN
Stun Servers

For Incoming calls:
Use WiFi
Use 3G
Use GPRS
Use EDGE
Use other networks

For Outgoing calls:
Use WiFi
Use 3G
Use GPRS
Use EDGE
Use other networks

Original comment by rene.s.h...@gmail.com on 26 May 2011 at 10:21

[GoogleCodeExporter]

Well, I tried r886 from trunk (May 26, 2011), and I still don't see anything in 
the expert Network settings that can let me set the start of the RTP port range.

Original comment by rene.s.h...@gmail.com on 26 May 2011 at 10:45

[GoogleCodeExporter]

Ah! Have to enable export mode for Settings from the Settings Menu. I was 
confused with export mode for Account editing: I though it affected settings 
too (and, of course, didn't bother to RTFL (read the fine link) provided.)

I presume RTP port 0 means "default" (4000-4007)?

Original comment by rene.s.h...@gmail.com on 26 May 2011 at 10:50

[GoogleCodeExporter]

Mmmm, RTP by default should be to 4000 not set to 0 (UDP and TCP are set to 0 
by default which mean random). 
On RTP, 0 (random) is not allowed.

Original comment by r3gis...@gmail.com on 28 May 2011 at 9:27