[Asterisk-Users] RE: iLBC

[Juha Heinanen]

after mark changed ilbc payload size to 50 bytes, */kphone started to
understand each other, but if i call from * to kphone, there is huge
delay.  if i call from kphone to *, voice quality is good from * to
kphone, but i can't hear anything from kphone (not even a ringing sounds
that may point to problems with oss, not ilbc).

if we forget the oss problems, then it remains to replace host address
in ruri/to/from headers with name.  an example from
draft-ietf-sipping-basic-call-flows-02.txt:

  F1 INVITE Alice -> Bob 
    
   INVITE sip:bob at biloxi.example.com SIP/2.0 
   Via: SIP/2.0/TCP client.atlanta.example.com:5060;branch=z9hG4bK74bf9 
   Max-Forwards: 70 
   From: Alice <sip:alice at atlanta.example.com>;tag=9fxced76sl 
   To: Bob <sip:bob at biloxi.example.com> 
   Call-ID: 3848276298220188511 at atlanta.example.com 
   CSeq: 1 INVITE 
   Contact: <sip:alice at client.atlanta.example.com;transport=tcp> 
   Content-Type: application/sdp 
   Content-Length: 151 
   ...

and some quotes from rfc 3261 (8.1.1 generating a request):

Request-URI The initial Request-URI of the message SHOULD be set to the
value of the URI in the To field.

To The To header field first and foremost specifies the desired  logical
recipient of the request, or the address-of-record of the user or
resource that is the target of this request.