Send a user defined String within the SDP packet - sip

Question

is there a way to send a short user defined string from the Caller to the Callee within the SDP part of an INVITE message (in a manner like steganography)? I tried to set the string with a length of approximately 15, in the k=, p=, e=, u= field. However the Asterisk server does not accept the Invite message. For sure, I set the new length in the IP-Header and UDP-Header furthermore I calculated the new Internet checksum of the IP-Header. As well I considered the CRLF scheme and the order of the fields.
Goal is, to transport data within the SDP data from the Caller to the Callee and vice versa, when the Callee responds with the 200 OK message to the Caller.
Thank you in advance!
Message with i=111.111.111.111 which is not accepted by the Asterisk:
INVITE sip:1000#192.168.0.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.2:6060;rport;branch=z9hG4bKGvBkM0qF4
Max-Forwards: 70
To: <sip:1000#192.168.0.14>
From: <sip:2000#192.168.0.14>;tag=SOXFP4ir
Call-ID: BEkXWRwn-1318101970419#x61.local
CSeq: 39 INVITE
Content-Length: 231
Content-Type: application/sdp
Contact: <sip:2000#192.168.11.2:6060;transport=UDP>
v=0
o=user1 1396633799 2096570444 IN IP4 192.168.11.2
s=-
i=111.111.111.111
c=IN IP4 192.168.11.2
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
Same Message but without i=111.111.111.111. This packet is accepted and the call proceeding ends successfully (with TRYING, RING 200OK)
INVITE sip:1000#192.168.0.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.2:6060;rport;branch=z9hG4bKESGSZD1V6
Max-Forwards: 70
To: <sip:1000#192.168.0.14>
From: <sip:2000#192.168.0.14>;tag=YPPrCWLp
Call-ID: 10MpKHYD-1318102031971#x61.local
CSeq: 41 INVITE
Content-Length: 211
Content-Type: application/sdp
Contact: <sip:2000#192.168.11.2:6060;transport=UDP>
v=0
o=user1 1682420165 643979666 IN IP4 192.168.11.2
s=-
c=IN IP4 192.168.11.2
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
Actually everything looks fine. And I can not see the response. I am intercepting the packets over iptables with NFQUEUE. Then just a few strstr, memcpy etc. to alter and build the new packets. I know there are some SDP stacks/APIs but in my case the quick and dirty solution is sufficient.

Answer 1

Try the i= field. As stated in RFC 4566:
The "i=" field is intended to provide a free-form human-readable description of the session or the purpose of a media stream. It is not suitable for parsing by automata.
So the Asterisk server shouldn't validate that field, enabling you to put the text you want there.