Problems with incoming VoIP 3.x calls

[edikojsi]

Hi fellows,

there is a very annoying problems with all VoIP Release 3.x phones – I've tried with E52, E55, E72, E75, N97, N97 mini.

If you register to ITSP normally everything is OK – you can make both outgoing and incoming VoIP calls.

The situation is different when you register a Nokia VoIP 3.x phone to an IP PBX, i.e. Siemens, Aastra, Panasonic, Asterisk and probably other SIP servers. You can make outgoing calls, but you an not receive any incoming call – you get either busy or other error.

If you look on the SIP messages you will see that the phone returns SIP error 407: Not found. Trying to debug with Wireshark you see that in some cases the SIP invite message is sent to another port (not 5060) for some reason.

This happens anywhere – in any 'test' lab, GPRS, etc., so it is not a problem of a specific LAN (router, DHCP) or IP PBX. It seems like the VoIP settings are fake.

Does anybody met the same problem or have any idea how to overcome it?

Best regards,
Edi.

[Chanflito]

I have the same problem.

Can I help me?

Thanks

[mreinder]

I have the same problem...

[Huwylersimon]

I have the same problem, already posted a new thread.

Which INVITE do you mean, is sent to another port, not 5060? The one to the Nokia? This one is not supposed to go to 5060 but to the contact port(as the Nokia is the client, which might, no... WILL be behind a PAT)

[edikojsi]

Hi,

this is a wireshark dump of N958GB and E52 in a same conditions. Both phones can make outgoing VoIP calls, but only N95 can receive VoIP calls. The problem is not related to PAT (I've tried wothout PAT).

If you register to ITSP – I've tried several – normally everything is OK – you can make both outgoing and incoming VoIP calls.

The situation is different when you register a Nokia VoIP 3.x phone to an IP PBX, i.e. Siemens, Aastra, Panasonic, Asterisk and probably some other SIP servers. You can make outgoing calls, but you an not receive any incoming call – you get either busy or other error.

If you look on the SIP messages you will see that the phone returns SIP error 404: Not found. Trying to debug with Wireshark (please see attached file and note below) you can see that in some cases the SIP invite message is sent to another port (not 5060) for some reason. There is no such problem on Nokia VoIP 2.x phones – like E51, E65, E66, E71, N95, etc.

This happens anywhere – in any 'test lab', GPRS, etc., so it is not a problem of a specific LAN (router, DHCP) or IP PBX. It seems like the VoIP settings are fake.

Please send me an e-mail address to edikojsi@yahoo.com to post you a wireshark capture or download it from:

http://4storing.com/0tpmu4/8105692ce...0fd73bf1b.html

Note that sip user '11' is the E52, '77' is the N95, '63' is another proprietary phone (Panasonic).
The invite message to E52 is sent to port 5060, while to N95 is sent to port 55002, which is working.

Kind regards,
Edi.

[Huwylersimon]

I think I found the problem!

When VoIP3.x registers a user, it writes a hash in the contact URI, i.e. "aweofj293j2c0239jdwiofje0@mydomain.org"
To invite a user, the server has to put this hash in the request-URI.

Asterisk seems to do so, iptel.org doesn't, neither does miniSipServer. I've put a proxy between the server and a client that remembers this hash for every user and copies this hash into the request-uri at every outgoing INVITE from the server, and now it works.

I don't know who is breaking the standard, but at the moment, there just seem to be servers that don't like Nokia Feature pack 2.

Greets
Simon

[rjschenk]

I think I found the problem!

When VoIP3.x registers a user, it writes a hash in the contact URI, i.e. "aweofj293j2c0239jdwiofje0@mydomain.org"
To invite a user, the server has to put this hash in the request-URI.

Asterisk seems to do so, iptel.org doesn't, neither does miniSipServer. I've put a proxy between the server and a client that remembers this hash for every user and copies this hash into the request-uri at every outgoing INVITE from the server, and now it works.

I don't know who is breaking the standard, but at the moment, there just seem to be servers that don't like Nokia Feature pack 2.

Greets
Simon

Good find.

Is there anywhere some official information by Nokia on this?

I'd like to contact a SIP provider to ask if they could fix the problem from their side.

Edit: Nokia VoIP Release 3.0 FAQ

1.32 Why does the Nokia S60 VoIP Release 3.0 product fail registration after receiving a ‘200 OK’ response from the registrar?
The Nokia S60 VoIP Release 3.0 product uses the pseudo-random username of the Contact URI for matching the sent and received contact(s). If the username part of the URI is changed or removed, the SIP registration of the Nokia S60 VoIP Release 3.0 product fails after the ‘200 OK’ response.
According to Nokia’s interpretation of the SIP RFC 3261, the registrar shall not modify the Contact URI sent in the response to the register request. When forwarding a request, for example, INVITE, some SIP servers write their own address to the Contact URI, instead of correctly inserting a Record-Route header. This also works with the Nokia S60 VoIP Release 3.0 product, but as for the REGISTER request the Contact has a different meaning and should not be touched.

Why do you like to break things for your customers, Nokia? :-(

[traud]

rjschenk, does this help?

[rjschenk]

rjschenk, does this help?

If the whole world does it one way and Nokia does it the other way, then Nokia is still breaking things for their customers, no matter if they're wrong or right.

[edikojsi]

The question is whether there will be a workaround, regardless Nokia or somebody else will provide it. It should be universal (i.e. work with most registrars), otherwise there is no much sense in thinking Nokia as SIP device.

[traud]

If the whole world does it one way and Nokia does it the other way, then Nokia is still breaking things for their customers, no matter if they're wrong or right.

How should a device with several VoIP/SIP clients onboard know how to redirect answers internally? In my understanding, that is the purpose of the contact header field. Anyway, does it matter? No. The trick is to patch your server and make it compatible. Zillions of users are happy about that, today. Do not sit and wait, do and change. If you do not have control over your server, find another one which works. Period.

[edikojsi]

How should a device with several VoIP/SIP clients onboard know how to redirect answers internally? In my understanding, that is the purpose of the contact header field. Anyway, does it matter? No. The trick is to patch your server and make it compatible. Zillions of users are happy about that, today. Do not sit and wait, do and change. If you do not have control over your server, find another one which works. Period.

This is possible, but I'm not sure whether it is the real reason. There are no such problems with multiple user registration on Nokia VoIP 2.0 devices, so why should there be any with VoIP 3.x.

It's a pity Nokia is not compliant, you can not change / patch the whole world to make Nokia SIP client working
Sorry, but your advice is not solving the problem.

[traud]

It's a pity Nokia is not compliant

Not compliant to what?

[N85 User]

Hi, i found this:

http://discussions.europe.nokia.com/...ighlight/true#

Really, i don't know what is the "realm" (at least in spanish).

[traud]

N85-User, in VoIP 3.x the realm is not important as it was in 2.x because now the realm is received (and re-used) from the first response of your server. Here in this thread, you have to patch your registrar/proxy (not to touch the contact header field) or place another proxy in between as Simon described. However, whether that hash in the contact header is the cause of your problem, should be checked with a packet analyser beforehand.