[Freeswitch-users] Incoming call audio problem

[David Knell]

Hi Brian,

With respect, it's not; it's sending a multipart message, of which the 
SDP (to which it assigns a
Content-Type: of application/sdp) is one.

RFC3261 says that multipart bodies are OK *if* the receiving end has 
said that it'll accept
them (I've appended the relevant bit); clearly there's a problem here if 
you're generating the
first message, as you don't know what the receiving end will accept.

The Cisco sends a multipart body, gets a 415 back with "Accept: 
application/sdp", and then
resends the INVITE with just the SDP as the body.  That all looks pretty 
reasonable.
FreeSWITCH then repeats the 415 message - presumably it's expecting it 
to be ACKed
in the same way as a 407, but the Cisco's not doing so.

Cheers --

Dave

7.4.1 Message Body Type

   The Internet media type of the message body MUST be given by the
   Content-Type header field.  If the body has undergone any encoding
   such as compression, then this MUST be indicated by the Content-
   Encoding header field; otherwise, Content-Encoding MUST be omitted.
   If applicable, the character set of the message body is indicated as
   part of the Content-Type header-field value.

   The "multipart" MIME type defined in RFC 2046 [11] MAY be used within
   the body of the message.  Implementations that send requests
   containing multipart message bodies MUST send a session description
   as a non-multipart message body if the remote implementation requests
   this through an Accept header field that does not contain multipart.

   SIP messages MAY contain binary bodies or body parts. When no
   explicit charset parameter is provided by the sender, media subtypes
   of the "text" type are defined to have a default charset value of
   "UTF-8".



> The cisco is sending the SDP as multipart/mixed which is retarded. 
>  This isn't supported by FreeSWITCH.
>
> /b
>
> On Mar 13, 2008, at 9:53 AM, Josip Djuricic wrote:
>
>> I can tell you that FS is returning 415 errors, not AS.
>>
>> When sofia is inviting everything is well, but when as sends request 
>> then the fun starts
>>
>> Brian West wrote:
>>> I can tell you this isn't the problem ... Sofia enables ip6 by default  
>>> on linux.  The AS5300 will spew out 415 errors like crazy.
>>>
>>> /b
>>>
>>> On Mar 13, 2008, at 9:21 AM, Tamas wrote:
>>>
>>>   
>>>> Hi,
>>>>
>>>> What kind of codec list do you use on as5300? Do you have maybe ACLs
>>>> defined?
>>>> Do you use the same dial-peers for incoming and outgoing calls too?  
>>>> What
>>>> IOS version do you run?
>>>> Could you share your cisco running-config?
>>>> Could you send also the SIP messages (without RTP)?
>>>>
>>>> Regards,
>>>>    Tamas
>>>>
>>>> Josip Djuricic írta:
>>>>     
>>>>> Hi there,
>>>>>
>>>>> I was already writing about this problem, and I still didn't  
>>>>> managed to
>>>>> fix it. I thought it was a NAT problem, but now without NAT the  
>>>>> problem
>>>>> still remains.
>>>>>
>>>>> The error is like this, when I call from pstn, the call goes trough
>>>>> Cisco AS5300, then goes to FS, then to it's final destination - Ip  
>>>>> Phone.
>>>>>
>>>>> Now you can answer the phone, but you can't hear anything on any  
>>>>> side.
>>>>> And FS returns lots of SIP 415 (unsupported media type) to Cisco  
>>>>> AS5300.
>>>>>
>>>>> When you call from Voip phone to the same pstn number it all works  
>>>>> well,
>>>>> so I'm starting to be confused more and more. (and in this case  
>>>>> there is
>>>>> no 415 messages).
>>>>>
>>>>> I believe cisco is sending something that FS can't understand, in the
>>>>> INVITE's but that's just a wild guess cause I ain't sure what can FS
>>>>> exactly understand very well.
>>>>>
>>>>> Whoever want's I can send him a tcpdump, it's about 7mb tarred, so I
>>>>> can't send it to the list.
>>>>>
>>>>> Thank you all,
>>>>>
>>>>> Josip
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>>
>>>>>       
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>>
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-- 
David Knell, Director, 3C Limited
T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
http://www.3c.co.uk 

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