#1023 Direct SIP calls - name resolutions

[Anonymous]

Originally created by: yok...@gmail.com

What steps will reproduce the problem?
1. Create account based on "Local" wizard
2. Make a SIP call to an account with "sip2sip.info" or "callcentric.com". To make direct SIP calls use format:
17772XXXXXX@in.callcentric.com
2233XXXXXX@sip2sip.info
(some digits are intentionally replaced with X)
3. Call immediately ends

What is the expected output? What do you see instead?
Ringing the destination phone, registered with corresponding account. Answering the call...

What version of the product are you using? On what operating system?
Latest version from the Market - v.0.02-01 [r841]. Android.

Please provide any additional information below.
CSS can't make any direct SIP calls (without registering with other SIP provider). At the same time exactly the same calls under the same conditions (no registration with any SIP provider, using the same format of caller info, etc) could be made using desktop PhonerLite program.

The problem is with DNS resolution mechanism, used in CSS program. Errors logged in 2 cases (calling "callcentric.com" and "sip2sip.info" correspondingly) are:
1. sip_resolve.c  Failed to resolve 'in.callcentric.com'. Err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
2. 403 Forbidden

Failure in the 1st case - there is no IP address, associated with the host "in.callcentric.com"
Failure is the 2nd case - "sip2sip.info" host used by CSS to send SIP INVITE doesn't support it (thus, "forbidden"). CSS should communicate with "proxy.sipthor.net" instead (and that host will not send "forbidden" reply)...

The proper procedure of name resolutions used by PhonerLite client in 1st case is:
1. Standard query SRV _sip._udp.in.callcentric.com
2. Standard query response SRV 20 0 5070 alpha5.callcentric.com
3. Standard query A alpha5.callcentric.com
4. Standard query response A 204.11.192.35

Similarly for sip2sip.info (2nd case):
1. Standard query SRV _sip._udp.sip2sip.info
2. Standard query response SRV 0 0 5060 proxy.sipthor.net
3. Standard query A proxy.sipthor.net
4. Standard query response A 85.17.186.7

Then SIP client should communicate with obtained IP's, while still using official host names, taken from provided SIP URI's:
1. 17772XXXXXX@in.callcentric.com
2. 2233XXXXXX@sip2sip.info

Currently CSS is using oversimplified name resolution procedure. Without this important fix (proper name resolution procedure) it's impossible to make direct calls to mentioned (as well, as others) SIP providers...

[Anonymous]

Originally posted by: r3gis...@gmail.com

Go in Settings > Network > Enable Resolve DNS SRV checkbox.
That's it :)

It's not activated by default cause on some old device it produce weird behavior, and also considerably reduce speed of startup of the app. But if you have a local account that's not a problem for you. Just activate that and it will work as you expect.

Status: WrongConfig

[Anonymous]

Originally posted by: yok...@gmail.com

That was the fastest fix, that I ever expected to see. In a minute all direct SIP calls start working. Thanks a lot! :D

Now, to make direct SIP calling mode usable and practical - there should be a way for that account to specify:
1. My name (CNAM)
2. My SIP phone number (to call me back)

It'd be nice to see those options in the "Local" wizard. Do you have any plans to offer that?

[Anonymous]

Originally posted by: r3gis...@gmail.com

It depends on the pjsip/pjsua library on which csipsimple relies.

For now, local accounts does not allow that (that's the only type of account seen differently by the sip stack cause the account id / contact has to be computed by the stack regarding the current ip).
So if they introduce a way to add a display name and an user name, csipsimple will benefit it and I'll add more settings in local account.

Or maybe I could patch pjsip to allow that (but usually I resent to do that cause things becomes less easy to maintain). ;)

Another workaround if you have a static IP (for example static lease in your wifi dhcp), could be to create an expert account without registrar and without proxy and with an account id : "Your name" <sip:XXXXX@YOUR_IP:YOUR_LOCAL_BOUNDED_PORT>

(:YOUR_LOCAL_BOUNDED_PORT is probably useless if you never changed it in expert setting mode, by default csipsimple use 5060 as soon as you have created one local account).

I never tried it but ~should~ theorically work.

[Anonymous]

Originally posted by: yok...@gmail.com

And it does work! Thanks a lot for the tip! :D

Now I can make direct calls and specify who I am and to what account call me back. It works both with static local IP (just for test), or with FQDN (for regular usage). For example:
"My name" <sip:1234@mySip.net>

I agree with you. CSipSimple would be definitely benefit in case if developers of pjsip/pjsua library included those two strings (please ask them about it). The logical place to configure those two additional fields (display name and URI to call back) is in "Local" account. BTW, I'd call it "Direct" call account, because that what it does - it makes a direct (and not via proxy SIP server) calls to a caller's SIP provider. But I can live with "Local" name too. ;)

Thank you again!