Request for Nokia experts! 415, unsupported media type

[hazmat]

Hi all,

Request:

This is my fourth day to work in solving this problem and it has become an absolute night mere for me.I'm requesting Nokia experts to pay attention on this thread.I'm explaing my problem once again in different way for better understanding.

Problem:

I’m having a problem at remote end when initiator sends an INVITE request including, m=audio 16384 RTP/AVP 98, RTP/AVP media type audio .Remote Party responses back “415, unsupported media type” and application not able to receive an INVITE request.

The capabilities that has been defined in SIP Resolved plug-in is that

"<SIP_CLIENT ALLOW_STARTING=\"YES\">\
<SIP_HEADERS>\
<ACCEPT_CONTACT value=\"*;mobility=mobile;media=audio\"/>\
<ALLOW_EVENTS value=\"telephone-event\"/>\
<ACCEPT value=\"audio/basic\"/>\
<ACCEPT value=\"application/sdp\"/>\
</SIP_HEADERS>\
<SDP_LINES>\
<LINE name=\"m\" value=\"application 0 TCP SIP\"/>\
<LINE name=\"m\" value=\"audio 30000 RTP/AVP 98\"/>\
<LINE name=\"a\" value=\"rtpmap:98 telephone-event/8000/1\"/>\
</SDP_LINES>\
</SIP_CLIENT>");

and the SIP message that includes INVITE request, SIP headers and SDP part of the SIP message is mentioned below. This packet is sent by initiator to remote party and in result we got sip response 415.

INVITE sip:5001@10.1.7.133 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.124;branch=z9hG4bKb8270604b89ee8351ccf693908b83607
Route: <sip:10.1.7.124;lr;transport=TCP>
Via: SIP/2.0/TCP 10.1.7.115:5060;branch=z9hG4bKul4lqk7cf5hc7amdag2pqpg;rport
From: <sip:5000@10.1.7.124>;tag=djilqk1gs9hc6pnoag2i
To: <sip:5001@10.1.7.124;user=phone>
Contact: <sip:5000@10.1.7.115;transport=TCP>
Allow: INVITE
Expires: 120
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 429
v=0
o=Nokia-SIPUA 63351281939131000 63351281939131000 IN IP4 10.1.7.115
s=-
c=IN IP4 10.1.7.115
t=0 0
m=application 0 TCP SIP
m=audio 16384 RTP/AVP 98
a=rtpmap:98 telephone-event/8000/1

[Nokia Ron]

A lot of Nokia people are on vacation but I'll try to bring this to the attention of someone that can help.

Ron

[hazmat]

Thanks Ron...

I really want to keep this thread moving until problem will be solved.This would be highly appreciable from all of us.

Thanks in advance,

Haroon Azmat

[bifihi]

Hi everybody,

I also got problems with the SIP Resolver Plugin API:

My SIP client application won't start up on the receipt of INVITE requests. Instead, the Nokia native SIP phone application starts. How can I bypass this application?

I found information in Nokia's technical library, which says that the native SIP phone application is registered to handle RTP/AVP and it is not possible to re-register for this media type. A unique Accept-Contact header (but the same RTP/AVP media type) should be used in the SIP message to resolve this. Which means adding a line like
<ACCEPT_CONTACT value="*;mobility="mobile";media="audio"" />
in the XML capabilities description. Am I right on this?

The SDK documentation says that аll the parameters listed in this line must match to an Accept-Contact header in an incoming SIP request if present.

Does this means that:
My application can receive requests only if they support exactly the same parameters in an Accept-Contact header field?
There is no way of calling my client from Eybeam, X-Lite,...? (most SIP Clients freely available don't support caller preferences!)

Is there any other way to bypass the native client or to deactivate/uninstall it?

Looking forward to hear from you!

[hazmat]

Hi,

As per my understanding about this work around that we should add this line as a header of SIP INVITE request not in the capabilities.But i'm wondering how do i send different types of payload,AMR/8000/1, telephone-event/8000/1,for audio?

I tried this work around BCZ on both sides i've my own aplication but still not able to recieve INVITE request.

Have you tried that?

Thnaks in advance,

Haroon

[bifihi]

As to my understanding the Accept-Contact header should be added in both the XML capabilities description of the client and the INVITE request. If they match the client will receive the request, though I haven't tried this yet.

My problem would be the Accept-Contact header in the INVITE request. I need my application to be reachable also from other SIP clients, such as Eyebeam, X-Lite,... and others. But as they don't support caller preferences there is no Accept-Contact header in the requests.

I would need a way to deactive/bypass/uninstall/overrule... the native Nokia client.......
Seems I really got stuck here.

Any idea? Anyone?

[hazmat]

Hi,

I found a thread during googling that we cant remove a native client.

I'm requesting senior engineer and Nokia experts to pay attension on this thread and get out of us from it.

It is really becoming a night mere for all of us.


Thanks in advance,

Haroon Azmat

[hazmat]

Hi,

I'm still struggling with this problem anyone who can have any clue,would be appreciable.


Thanks in advance,

Haroon Azmat

[nly]

Element "<LINE name=\"a\" value=\"rtpmap:98 telephone-event/8000/1\"/>\" in the original XML is invalid.
Here is the corrected XML:

<SIP_CLIENT ALLOW_STARTING=\"YES\">\
<SIP_HEADERS>\
<ACCEPT_CONTACT value=\"*;mobility=mobile;media=audio\"/>\
<ALLOW_EVENTS value=\"telephone-event\"/>\
<ACCEPT value=\"audio/basic\"/>\
<ACCEPT value=\"application/sdp\"/>\
</SIP_HEADERS>\
<SDP_LINES>\
<LINE name=\"m\" value=\"application 0 TCP SIP\"/>\
<LINE name=\"m\" value=\"audio 30000 RTP/AVP 98\">\
<MEDIA_ATTRIBUTE value=\"rtpmap:98 telephone-event/8000/1\"/>\
</LINE>\
</SDP_LINES>\
</SIP_CLIENT>

[bifihi]

Thanks for that.

Could you please answer my question too:
Is there any way of getting my client started when calling from Eyebeam, X-Lite,... (no caller preferences, no Accept-Contact in INVITE request)?
Is the Accept-Contact header the only way to distinguish from the Native Nokia Client?

Thanks in advance!

[hazmat]

Hi,

I defined the capabilities as you mentioned but my application crashed when i tried to create a ACCEPT-CONTACT header like this...

_LIT8( KAcceptContactParam,*;mobility=mobile;media=audio)
accContactHeader = CSIPAcceptContactHeader:ecodeL( KAcceptContactParam() );

I'm supposing that Accept-CONTACT header should be part of INVITE message.I traced the reason of application crashing was mobility=moobile param.SO please give me clue how can i include this param in the header?


Apart from that I've added the ACCEPT-CONTACT header without including mobility param and removed that from Capabilities and from ACCEPT-CONTACT header of SIP INVITE message.I'm copying XML and INVITE message trace below please have a look and let me know why i'm getting
"480 temporary not available".

If adding of mobility is necessary then let me know how can i'm able to include this into SIP header.

<SIP_HEADERS>\
<ACCEPT_CONTACT value=\"*;media=audio\"/>\
<ALLOW_EVENTS value=\"telephone-event\"/>\
<ACCEPT value=\"audio/basic\"/>\
<ACCEPT value=\"application/sdp\"/>\
</SIP_HEADERS>\
<SDP_LINES>\
<LINE name=\"m\" value=\"application 0 TCP SIP\"/>\
<LINE name=\"m\" value=\"audio 30000 RTP/AVP 98\">\
<MEDIA_ATTRIBUTE value=\"rtpmap:98 telephone-event/8000/1\"/>\
</LINE>\
</SDP_LINES>\
</SIP_CLIENT>


SIP Invite Request trace......


INVITE sip:5000@10.1.7.124 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.118;branch=z9hG4bK7d274a13588368dce62f0e08cecfe824
Via: SIP/2.0/UDP 10.1.7.123:5060;branch=z9hG4bKpglimdhq41hc7aco4b4j6g1;rport
From: <sip:5001@10.1.7.118>;tag=uvd2mdn6ndhc6ugh4b4r
To: <sip:5000@10.1.7.118>
Contact: <sip:5001@10.1.7.123>
Accept-Contact: *;media=audio
CSeq: 986 INVITE
Call-ID: UZMHdEVmoIfXs2b9O70w3bPQXAIO8m
Allow: INVITE
Max-Forwards: 69
Content-Type: application/sdp
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 181

v=0
o=5000 188870569 2890844532 IN IP4 10.1.7.123
s=SipSession
c=IN IP4 10.1.7.123
t=0 0
m=application 0 TCP SIP
m=audio 30000 RTP/AVP 98
a=rtpmap:98 telephone-event/8000/1

[hazmat]

Please give me clue on this problem ...i'm almost on the way to implement RTP Framwork.To test my application i really need to remove this road block.I'm stuck at this point for last two, weeks please help me out.


Thanks in advance,

Haroon Azmat

[tkot]

Hi Hazmat!

Your problem lies (atleast) in that accept-contact. Latest SIP
versions require '+' sign for non-base-tags (which are listed in
rfc 3840, chapter 9). So, these should work:
<ACCEPT_CONTACT value=\"*;+media=audio\"/> OR
<ACCEPT_CONTACT value=\"*;audio\"/>

Naturally, you should have exact same value in your INVITE
in order to get match.

Hope this helps!

[hazmat]

Thanks alot,It works now.

Well done,I'm appreciated your gr8!!! reply.

So I'm closing this thread successfully.

Haroon Azmat

[zzhehai]

hazmat:
How do you solved this problem?
Can you tell us !

My KCapabilitiesis is follow,but it didn't receive the INVITE

_LIT8(KCapabilities,
"<SIP_CLIENT ALLOW_STARTING=\"YES\">\
<SIP_HEADERS>\
<ACCEPT_CONTACT value=\"*;+media=audio\"/>\
<ALLOW_EVENTS value=\"telephone-event\"/>\
<ACCEPT value=\"audio/basic\"/>\
<ACCEPT value=\"application/sdp\"/>\
<ACCEPT value=\"text/plain\"/>\
</SIP_HEADERS>\
<SDP_LINES>\
<LINE name=\"m\" value=\"audio 8000 RTP/AVP 18 8 0 101\" />\
<LINE name=\"m\" value=\"audio 18008 RTP/AVP 96 3 107 8 0 101\"/>\
<LINE name=\"m\" value=\"application 0 UDP mySIP\"/>\
<LINE name=\"m\" value=\"audio 30000 RTP/AVP 98\">\
<MEDIA_ATTRIBUTE value=\"rtpmap:98 telephone-event/8000/1\"/>\
</LINE>\
</SDP_LINES>\
</SIP_CLIENT>");

thanks!