[Asterisk-Users] Early Media in 100 Ringing

[Ronald Voermans]

Hello,

I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:

As you can see below, the SIP message from 10.254.254.1 (the PSTN
Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content.

How can this be solved?

U 10.254.254.1:5060 -> 192.168.0.173:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35.
Record-Route: <sip:0161801019 at 10.166.38.108:5060>.
Record-Route: <sip:0161888874 at 10.254.254.1:5060;lr;nat=yes>.
From: "0161801019" <sip:0161801019 at 192.168.0.173>;tag=as02de1b95.
To: <sip:0161888874 at 10.254.254.1>;tag=00-04094-52dbe3bc-6cf68a723.
Call-ID: 71f7297e0e6cc0625bbae5be00f8a2cc at 192.168.0.173.
CSeq: 102 INVITE.
Contact: <sip:212.241.48.70:5060>.
server: Cirpack/v4.38f (gw_sip).
Allow: UPDATE, REFER.
Content-Type: application/sdp.
Content-Length: 253.
.
v=0.
o=cp10 112775383044 112775383045 IN IP4 10.166.38.109.
s=SIP Call.
c=IN IP4 10.254.254.1.
t=0 0.
m=audio 35058 RTP/AVP 18 101.
b=AS:64.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.
a=ptime:20.

#
U 192.168.0.173:5060 -> 192.168.1.103:5062
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265.
From: "411" <sip:411 at 192.168.0.173>;tag=f93ee2f65c6906cb.
To: <sip:0161888874 at 192.168.0.173>;tag=as675f246d.
Call-ID: 56dc51e7f5084d4b at 192.168.1.103.
CSeq: 60590 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: <sip:0161888874 at 192.168.0.173>.
Content-Length: 0.
.