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Grammar:
* Grammar is not fully converted, see comments inside the grammar file
* UTF8 things not defined

Request/Response:
* Parse more Parameters properly...

Transaction:
* Cancel a Invite Transaction, stop responding to the 200.. or do it
  with a bye...
* Invite can have multiple dialogs.. so we need to have a 'new dialog'
  kind of signal...
* Route signals through the ok/cancel of the SIPDialog as a single entry
* Add responses...
* Add better state machine control, (do not allow to go back in states)


Dialog/Session/Route-Set:
* A dialog should have a route set (with the Via's)
* Does a dialog hold a session? a session holds a dialog?
* A call can go to session without having a confirmed dialog. This needs
  to be looked at/denied. Do not ack a call that has no ;tag= attribute
  for. One can remove the from ;tag in the "200 OK" test result to provoke
  the failure in the testProxyAuth testcase.

General:
* 401 handling might not work for BYE,ACK. For ACK the ACK might not
  be re-generated and the digest might be wrong due the missing operation
* 403 (maybe 401) should generate ACK on the 403 result? Check the RFC
  if that is necessary! This would indicate why the branch is changing
  _after_ the receiving 401/403. At the same time the proxy/server would
  need to re-send the 401/403?
* Compare with MGCPCommands and share code... in SIPRequest
* Via we should indicate the received address and port...
* Verify that status code is valid. It used to be done inside the
  grammar but that is not the right place.


* 3xx, 4xx, 5xx, 6xx are final. We should not allow any other messages.
Sending multiple 503/500 messages.. they all will be acked..


* Record-Route/Route implementation needed:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.251:3107;branch=z9hG4bK-qvgn1b;rport=3107
Record-Route: <sip:172.20.40.8;lr>
Record-Route: <sip:217.10.68.147;lr;ftag=7kypvq>
From: "sipgate" <sip:ABC@sipgate.de>;tag=7kypvq
To: "00491778116745" <sip:0049123@sipgate.de;user=phone>;tag=as10fec2f5
Call-ID: n57q3hl9l7
CSeq: 1084794278 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:49123@217.10.77.114:5060>
Content-Type: application/sdp
Content-Length: 381

v=0
o=root 1250091314 1250091315 IN IP4 217.10.77.114
s=sipgate VoIP GW
c=IN IP4 217.10.77.114
t=0 0
m=audio 17974 RTP/AVP 8 0 3 18 2 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

ACK sip:123@217.10.77.114:5060 SIP/2.0
Via: SIP/2.0/UDP 24.134.59.157:3107;branch=z9hG4bK-2tacy7;rport
Route: <sip:217.10.68.147;lr;ftag=7kypvq>
Route: <sip:172.20.40.8;lr>
From: "sipgate" <sip:ABC@sipgate.de>;tag=7kypvq
To: "00491778116745" <sip:0049123@sipgate.de;user=phone>;tag=as10fec2f5
Call-ID: n57q3hl9l7
CSeq: 1084794278 ACK
Max-Forwards: 70
Contact: <sip:abc@24.134.59.157:3107;transport=udp;line=i4p4q6>;reg-id=1;+sip.instance="<urn:uuid:572d1aa1-bfd5-4b8a-ab1e-f9095df386e5>"
Content-Length: 0



Reply with ACK to 401 authorized..

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK4000ebad-1c1f-e411-87f8-844bf52a8297;rport=1029;received=91.66.224.184
From: <sip:5100310@8.8.8.8>;tag=e698dead-1c1f-e411-87f8-844bf52a8297
To: <sip:00123@8.8.8.8>
Call-ID: 4eaedead-1c1f-e411-87f8-844bf52a8297@xiaoyu
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="Yate", nonce="cc615df915f40a7d8ae80ef28c8efba4.1407689697", stale=FALSE, algorithm=MD5
Server: YATE/5.4.0
Contact: <sip:00123@8.8.8.8:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0



ACK sip:00123@8.8.8.8 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.0.106:5060;branch=z9hG4bK4000ebad-1c1f-e411-87f8-844bf52a8297;rport
From: <sip:5100310@8.8.8.8>;tag=e698dead-1c1f-e411-87f8-844bf52a8297
Call-ID: 4eaedead-1c1f-e411-87f8-844bf52a8297@xiaoyu
To: <sip:00123@8.8.8.8>
Content-Length: 0
Max-Forwards: 70