[asterisk-users] RTP / NAT question ( pjsip )

Kevin Long kevin.long at haloprivacy.com
Wed Mar 2 18:52:49 CST 2016



I am having trouble with RTP and NAT :


Below is a SIP SDP invite from a remote endpoint which is trying to call extension 420 which is the ECHO application .


As you can see, the public IP is where the request comes in from,  but the SDP contains the private, internal IP in numerous places.


I do have rewrite_contact=yes;  on in my pjsip endpoint configuration,  but still the “rtp set debug on” command is showing me that when I dial into the echo application,  RTP packets are being sent to the private IP and not the public IP .



Advice appreciated thank you. 



<--- Received SIP request (1282 bytes) from TLS:72.52.31.109:55256 --->
INVITE sip:420 at dev1.domain.com SIP/2.0
Via: SIP/2.0/TLS 10.128.30.239:55253;branch=z9hG4bK-524287-1---bf28eb29eb900b43;rport
Max-Forwards: 70
Contact: <sip:6000 at 72.52.31.109:55256;transport=TLS;rinstance=e652ef90f2843e40>
To: <sip:420 at dev1.domain.com>
From: "Kevin"<sip:6000 at dev1.domain.com>;tag=0af40611
Call-ID: MGE5OWFhMDY5OGFhYzM4ZDIxNjA5OGRjY2M5OWE3ZGY
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, MESSAGE, OPTIONS, SUBSCRIBE, OPTIONS
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.6.2 stamp 33024
Authorization: Digest username="6000",realm="asterisk",nonce="1456965577/29f2977e5352209d33847b1eafc5f937",uri="sip:420 at dev1.haloprivacy.com",response="9c23bba47f43fa343bfc3bd2580a84ad",cnonce="ea996236e91c869bb16b1652c8504ba3",nc=00000001,qop=auth,algorithm=md5,opaque="609ab4014ccfac10"
Content-Length: 358

v=0
o=- 1456965576139402 1 IN IP4 10.128.30.239
s=Cpc session
c=IN IP4 10.128.30.239
t=0 0
m=audio 61216 RTP/SAVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:tkUxPSw8qTZ25fk6VuQPWNVOABk5mwe63/+d7vP7
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:tkUxPSw8qTZ25fk6VuQPWNVOABk5mwe63/+d7vP7
a=sendrecv

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