24 Nov
2014
24 Nov
'14
12:41 p.m.
Hello,
I don't understand what you want to do, however, if you look to change
content of the sdp, see the nathelper, rtpproxy, mangler, sdpops,
textops, textopsx and siputils modules -- these should give you plenty
of options to play with it to update the sdp.
Cheers,
Daniel
On 20/11/14 18:21, Javier Ricke wrote:
------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 <http://152.74.21.12:5060> --->
ACK sip:1001@152.74.21.12:6112 <http://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
<mailto:sip%3A1001@152.74.21.12>;transport=UDP>;tag=as6e487bf1
From: <sip:JavierTren@152.74.21.12
<mailto:sip%3AJavierTren@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
<mailto:sip%3A1001@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
<http://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
<http://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
<mailto:sip%3A1001@152.74.21.12>;transport=UDP>;tag=as6e487bf1
To: <sip:JavierTren@152.74.21.12
<mailto:sip%3AJavierTren@152.74.21.12>;transport=UDP>;tag=2b8fa52c
Contact: <sip:1001@152.74.21.12:6112 <http://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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--
Daniel-Constantin Mierla
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