Configuration of Nokia client SIP port

[chschaef]

we intend to using Nokia smartphones in our enterprise communication system which relies on SIPS (SIP+TLS) & SRTP (suing OpenSER)
- thus the user & registrar address is specifiied with the sips-uri and the registrar port number is set to 5061
- this allows to make a call from the Nokia to some other device fine
> however, when calling the Nokia, on port 5061, it does not respond and seems only to listen for connection request on port 5060: for using TLS, the expected value is 5061
> the expected behavior is that the Nokia listens on the same port as the SIP server

>> what workarounds are possible (besides special Nokia rulkes on OPENser)
>> what plans are there to correct this bug?

[mayankkedia]

Welcome to the Dibo :-)

Your query looks more like an end user one and for a more suitable and end-user focused answer please see http://www.nokia.com/support or http://www.nokia.com/discussions

[chschaef]

this question is here in the developer forum on purpose:
in slide set SIP_and_RTP_SRTP_V1.0_en.pdf, Paulius Meskauskas states that incoming TLS connections are not yet supported (that is not even part of the VoIP Release Notes)

for outgoing calls the Nokia uses port 5061 & TLS & TCP/IP
for incoming calls the Nokia uses port 5060 & UDP

>> anyone in the community or the Nokia developers if that has been re-solved?

thanks, Christoph

[meskausk]

Hello,

Which Nokia device are you using?
And could you pls post the SIP profile settings.

cheerio,
paulius

[chschaef]

we have been using an E71 with the following settings:

Nokia Configuration/SIP-Profile:
profile name: test (some name)
service profile: IETF
default access point: (the name of the WLAN AP)
public user name: sips:3608233@{MySIPSDomain} for TLS
compression: none
registration: on demand
security: off
proxy server: not defined
registration server:
address: sips:{MySIPSDomain} for TLS
realm: {MySIPSDomain}
user name: 3608233
password: {SIP password}
transport: AUTO for TLS
port: 5061 for TLS
in a configuration menue not accessable on the UI, SECURECALLPREF is set to 2 for SRTP
// security is mandatory for call establishment

Nokia Configuration/Web-Phone:
name: stest (some name)
SIP-profile: stest

Nokia Application/Communication Internet Phone:
default call type: Internet
settings for stest
registration: manual (vs. automatic)


(you can get the settings as an XML file, too)

[meskausk]

Settings look fine to me. Few questions

- Did you try to call E71 from E71 or using some other client?
- OpenSer forwards the INVITE to the port 5061?

We might have an issue with the offer received in case non-Nokia client
was used for MO call...

cheerio,
paulius

[chschaef]

we tried it out from another client (softphone).
The OpenSER uses port 5061 for the signalling (TLS).

[meskausk]

I reckon E71 simply does not understand the offer received and ignores it.
Are you able to place a call from E71 to another E71?
Could you post the offer OpenSER-server forwards to E71?

thanks,
paulius

[chschaef]

Calling an E71 from an E71 or an E51 fails.
We observed that the E71 seems not to accept INVITES on port 5061, i.e. seems to refuse the TCP connect.

The following is a SIP log from the call server (we have switched to Kamillio in the meantime):
(as a substitute for a SIP over TLS trace on the Ethernet; please note that this is apparently not received by the E71.)

Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: ******************************SIP MESSAGE start**********************
Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: INVITE sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.14.96.86:5060;branch=z9hG4bK2eh6u021aitebpato1dh9dr;rport
From: <sips:3608233@fe-ucs-d03.fe.de.bosch.com>;tag=4srs0mu85hhc6f8qvn1n
To: <sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone>
Contact: <sips:3608233@10.14.96.86>
Supported: 100rel,precondition
CSeq: 1614 INVITE
Call-ID: FKP0mVaeoIcqg_T3K-1q4d3QQxh82f
Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK
Expires: 120
Privacy: none
User-Agent: E71-1 RM-346 200.21.118
P-Preferred-Identity: sips:3608233@fe-ucs-d03.fe.de.bosch.com
Max-Forwards: 70
Proxy-Authorization: Digest realm="10.51.2.6",nonce="4a6416fb00005085da185b2ba8fbd6c9a78ccaa5827760e4",algorithm=MD5,username="3608233",uri="sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone",response="dee8e418b47ee566a0c1a3dc20e8a018"
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 606

v=0
o=Nokia-SIPUA 63416358214921250 63
Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: ***************************** SIP MESSAGE end************************
Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: [LOGMSG] [INVITE] from [3608233] to [3608234]
Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: ******************************SIP MESSAGE start**********************
Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: ACK sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.14.96.86:5060;branch=z9hG4bK2eh6u021aitebpato1dh9dr;rport
From: <sips:3608233@fe-ucs-d03.fe.de.bosch.com>;tag=4srs0mu85hhc6f8qvn1n
To: <sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone>;tag=ef496c44245079ef90755c12b7ab698c-b886
Call-ID: FKP0mVaeoIcqg_T3K-1q4d3QQxh82f
CSeq: 1614 ACK
Supported: sec-agree
Max-Forwards: 70
Proxy-Authorization: Digest realm="10.51.2.6",nonce="4a6416fb00005085da185b2ba8fbd6c9a78ccaa5827760e4",algorithm=MD5,username="3608233",uri="sips:3608234@fe-ucs-d03.fe.de.bosch.com;user=phone",response="dee8e418b47ee566a0c1a3dc20e8a018"
Content-Length: 0


Jul 20 09:03:58 fe-ucs-d03 /usr/local/sbin/kamailio[30303]: ***************************** SIP MESSAGE end************************

[strutv]

Hi chschaef

Do you use Native Nokia Internet Telephony app to make in/out calls?
And did you install some server certificate to the phone to make TLS connection with server?
I'm trying to register SIP profile first of all according to TLS settings provided by you above, but receive Registration failed.

Thanks
*strutv

[chschaef]

on the Nokia, the CA certificate for the Call Server is installed but that depends on your call Server configuration
best have a look in the OpenSER log files on what is wrong

Christoph