[SIPForum-discussion] SIP Testing (INVITE without SDP)

Mon Mar 17 21:52:11 UTC 2008

Sam,

We are doing the fast start as the Inbound INVITE is with SDP and the
INVITE for redirecting media is without SDP

INBOUND INVITE

<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

INVITE sip:3033300990 at 172.244.11.13:5065;user=phone SIP/2.0

v: SIP/2.0/UDP
172.244.11.13:5060;branch=z9hG4bK0c8d9fb44874689ee1932ae4cd185eed.1,SIP/
2.0/UDP 172.244.11.13:5060;branch=z9hG4bKok3pjm1070k1gdc8s640.2

f: "WIRELESS CALLER"
<sip:7202730836 at 6192.168.23.13:5060>;tag=6d07962cb6fc434af56dd301614776d
1

t: <sip:3033300990 at x.x.x.x:5060>

i: 3765ad045e80d167852417d8d84cf4ee-47d845e4 at 192.168.23.13

CSeq: 11890 INVITE

Max-Forwards: 68

Allow:
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,OPTIONS

m: "WIRELESS CALLER"
<sip:SD1fhg9-jtrl8c167i4qf163fov98164103k2v37vl0vd4bhvegjg84opovjm316s7@
192.168.23.13:5060;transport=udp>

Remote-Party-ID:   "WIRELESS CALLER"
<sip:7202730836 at 192.168.X.X:5060>;privacy=off

k: timer,100rel

x: 1800

Min-SE: 1800

l: 287

c: application/sdp

v=0

o=Sonus_UAC 25600 2560000 IN IP4 192.168.23.13

s=SIP Media Capabilities

c=IN IP4 192.168.23.13

t=0 0

m=audio 46324 RTP/AVP 18 0 100

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:100 telephone-event/8000

a=fmtp:100 0-15

a=sendrecv

a=maxptime:20

<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< 

OUTBOUND INVITE (for redirect call)

>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

INVITE sip:3013442660 at 192.168.23.13:5060 SIP/2.0

From: "WIRELESS CALLER"
<sip:7202730836 at 172.244.11.13>;tag=10349ae2-1dd2-11b2-bc05-b03162323164+
1a6e99f4

Contact: "WIRELESS CALLER"
<sip:7202730836 at 172.244.11.13:5065;transport=udp>

To: <sip:3033300990 at x.x.x.x>

Call-ID: 950-452333452 at 192.168.23.13:

CSeq: 2 INVITE

Max-Forwards: 70

Allow: INVITE,BYE,ACK,CANCEL,PRACK,REFER,OPTIONS,REGISTER,NOTIFY

Diversion:
<sip:3033300990 at 72.245.116.137>;reason=no-answer;privacy=off;counter=3

Remote-Party-ID: "7202730836"
<sip:7202730836 at 192.168.23.13:>;party=calling;id-type=subscriber;privacy
=off

P-Asserted-Identity: "7202730836" <sip:7202730836 at lab.mytest.com>

Diversion:
<sip:3033300990 at 192.168.23.13:>;reason=no-answer;counter=3;privacy=off

Via: SIP/2.0/UDP 192.168.23.13::5065;branch=z9hG4bK94550788724048

Content-Length: 0

>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

Regards,

Rashid Shakil

-----Original Message-----
From: Chong Fah Sam [mailto:fahsam at xener.com] 
Sent: Friday, March 14, 2008 4:45 AM
To: Shakil, Rashid; discussion at sipforum.org
Subject: RE: [SIPForum-discussion] SIP Testing (INVITE without SDP)

Hi, 

This is because the inbound call is using the slow start (INVITE without
SDP). And also your softswitch is not supporting the delay offer (INVITE
without SDP).

The solution is either you can configure your softswitch to support the
slow start or request the incoming call enable the fast start.

Best Regards,

Sam 

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Shakil, Rashid
Sent: Thursday, March 13, 2008 10:18 PM (TT) 
To: discussion at sipforum.org
Subject: [SIPForum-discussion] SIP Testing (INVITE without SDP)

Hello All,

I am having call failure issues when my softswitch sending a SIP INVITE
without SDP for a redirect call (Inbound call from SIP redirected back
to an offnet number). The call scenario is as follows

1.      An Inbound call (PSTN to IP) comes in from SIP carrier. The user
on SIP side (our customer) enable the Find me feature to redirect the
inbound to an offnet number

2.      An outbound call goes out from my softswitch to the SIP carrier.
The INVITE for that outbound call has diversion header but no SDP

3.      In response to the outbound INVITE (without SDP) the upstream
Gateway sent me 480 Temporary Unavailable with this explanation "After
receiving the PSTN call, the gateway sends out the first INVITE. It
receives back the forwarded INVITE (with diversion header), but the
INVITE does not have any media specified. So the gateway then sends out
a new INVITE to the destination number, but sets media a=sendonly. The
call is being rejected because on the egress side, the INVITE goes out
with a=sendonly (because there was no SDP on the Ingress INVITE), but
the 200 OK comes back with no "a" setting, which means a=sendrecv".

I believe we can redirect a call with diversion header and without SDP.
Is this explanation from my upstream provider is correct do I have to
send an SDP for the redirected call... Please reply me so that I can
move forward with this issue. 

Regards,

Rashid Shakil.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20080317/6ea6f54b/attachment-0002.html>