[Asterisk-Users] Early Media in 100 Ringing

[Joshua Colp - Asterlink]

Hello Ronald,

A 180 Ringing is something that should not have SDP because it's out of band
signaling of the exact status of the call, ringing. The PSTN Gateway should
return a 183 Session Progress if it wants to deliver inband audio progress.
Their SIP implementation doesn't look the best either... so to get it to
work you'd either have to hack Asterisk, or get the manufacturer of the PSTN
gateway to fix their stuff.

Joshua Colp

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ronald
Voermans
Sent: Monday, September 26, 2005 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Early Media in 100 Ringing

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.
.
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