Dear all:
I'm new in SIP. I'm trying to develop a softphone. I'm testing my softphone 
with sip.inphonex.com, when I make a call this is the resulting dialog:

MySP ---> Remote
INVITE sip:[email protected] SIP/2.0
Call-ID: [email protected]
From: <sip:[email protected]>;tag=12345
To: <sip:[email protected]>
Via: SIP/2.0/UDP 
10.168.123.12:5060;branch=z9hG4bKad65cbbd6fc59718bc5ffb49f00fc0c2
Max-Forwards: 70
Contact: <sip:[email protected]:5060;lr>
Allow: ACK,INVITE,BYE,CANCEL,SUBSCRIBE,NOTIFY
Content-Type: application/sdp
CSeq: 2 INVITE
Proxy-Authorization: Digest 
uri="sip:[email protected]",realm="sip.inphonex.com",username="2427000",nonce="49e626477a8b02497d6b9efa7bc36d908d69beb5",algorithm=MD5,response="1c52970803f10d9dd45f2a03eb5d352b"
Content-Length: 234

v=0
o=- 6 2 IN IP4 10.168.123.12
s=Syrium-Softphone
c=IN IP4 10.168.123.12
t=0 0
m=audio 57724 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

Remote ----> MySP

SIP/2.0 180 Ringing
Call-ID: [email protected]
From: <sip:[email protected]>;tag=12345
To: <sip:[email protected]>;tag=SDeivvf99-16146df4
Via: SIP/2.0/UDP 
10.168.123.12:5060;received=10.168.121.200;branch=z9hG4bKad65cbbd6fc59718bc5ffb49f00fc0c2;rport=12894
CSeq: 2 INVITE
X-UUID: fdd6e622cc00449ab4953e6cbb8efc31
Content-Disposition: session;handling=required
Content-Type: application/sdp
Allow: INVITE
Content-Length: 181

v=0
o=Sonus_UAC 15634 19687 IN IP4 208.239.76.169
s=SIP Media Capabilities
c=IN IP4 208.239.76.169
t=0 0
m=audio 50106 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20


I have two doubts:


        * Which ports must comunicate with which ports?
        * 208.239.76.169:50106   ----> 10.168.123.12:50106 and 
208.239.76.169:57724   ----> 10.168.123.12:57724
        * or what?
        * Is something wrong with the messages? This question is because I 
can't receive or sent any audio stream.Thanks for your help, and sorry for the 
bad english.



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