------phona A-------kamailio---------asterisk-----

OPTION 1: configure asterisk or kamailio

i used asterisk, and install kamailio for traffic RTP can be send between end points that behind NAT router, and do not have to go through RTP proxy,, plz help!

i think to the moment install kamailio, headrs'sdp fix IP private, but no!, how can fix it plz!!?

regards!

or 

OPTION 2: edit sip/sdp

mi sip/sdp is

[code]

<--- SIP read from UDP:152.74.21.12:5060 --->

ACK sip:1001@152.74.21.12:6112 SIP/2.0

Via: SIP/2.0/UDP 152.74.21.12;branch=z9hG4bKcydzigwkX

Via: SIP/2.0/UDP 190.164.204.227:41553;branch=z9hG4bK-d8754z-84f73c7e042445de-1---d8754z-;rport=41553

Max-Forwards: 16

Contact: <sip:JavierTren@190.164.204.227:41553;transport=UDP>

To: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1

From: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c

Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.

CSeq: 2 ACK

Proxy-Authorization: Digest username="JavierTren",realm="152.74.21.12",nonce="VGFnblRhZkIpJRpScSaEi795VKe4uof0",uri="sip:1001@152.74.21.12;transport=UDP",response="116bf459c22231d0a770534d674b768d",algorithm=MD5

User-Agent: Zoiper r27147

Content-Length: 0

<------------->

--- (12 headers 0 lines) ---

set_destination: Parsing <sip:152.74.21.12;lr=on;ftag=2b8fa52c> for address/port to send to

set_destination: set destination to 152.74.21.12:5060

Audio is at 12064

Adding codec 100002 (gsm) to SDP

Adding codec 100003 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 152.74.21.12:5060:

INVITE sip:JavierTren@190.164.204.227:41553;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 152.74.21.12:6112;branch=z9hG4bK3992226d;rport

Route: <sip:152.74.21.12;lr=on;ftag=2b8fa52c>

Max-Forwards: 70

From: <sip:1001@152.74.21.12;transport=UDP>;tag=as6e487bf1

To: <sip:JavierTren@152.74.21.12;transport=UDP>;tag=2b8fa52c

Contact: <sip:1001@152.74.21.12:6112>

Call-ID: NjgyMTViMDAyMzczNjIyNWIwZWU3OWJjZDAxMWFkNjY.

CSeq: 102 INVITE

User-Agent: Asterisk PBX 11.13.0

Session-Expires: 1800;refresher=uac

Min-SE: 90

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

X-asterisk-Info: SIP re-invite (External RTP bridge)

Content-Type: application/sdp

Content-Length: 256

v=0

o=root 1842142539 1842142540 IN IP4 192.168.1.8

s=Asterisk PBX 11.13.0

c=IN IP4 192.168.1.8

t=0 0

m=audio 8000 RTP/AVP 3 0 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

[/code]

and the field o and c i need to IP public y no private,,

plz any?

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