Voice gateway, SIP Options ping without SDP

[Oleh Doroshenko]

Hi all.

I see following SIP options call-flow for sip options ping:

17907581: Jan 16 08:11:31.057: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.250.20.25:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
Call-ID: c7vb43afudub9c4f944edcf4a937p4s3@SBC.com
From: <sip:SBC@10.11.3.17>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

17907583: Jan 16 08:11:31.057: //28111388/1CCF073F9ED3/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
From: <sip:SBC@10.11.3.17>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>;tag=BC08FBDC-1F16
Date: Fri, 16 Jan 2015 08:11:31 GMT
Call-ID: c7vb43afudub9c4f944edcf4a937p4s3@SBC.com
Server: Cisco-SIPGateway/IOS-15.4.20141104.060737.
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 363

v=0
o=CiscoSystemsSIP-GW-UserAgent 1616 9170 IN IP4 10.13.4.43
s=SIP Call
c=IN IP4 10.13.4.43
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.13.4.43
m=image 0 udptl t38
c=IN IP4 10.13.4.43
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

I know that this is normal behavior for voice gateway(3925 in our case).

Is it possible to disable SDP in 200 OK message going out to ITSP. They said that their PBX can't recognize 200 ok with SDP as reply to sip Options message.

[Zekeria Sammantar]

Hi Oleh,

Please post the complete call flow debug. 

[Oleh Doroshenko]

This is complete SIP OPTIONS ping dialog. There is no problem with call establishment when options ping disabled but when ITSP cannot recognize options ping reply they consider host down and even do not initialize SIP call to that host.

[Ayodeji Okanlawon]

Oleh,

I am not sure you can do this..

RFC3261 states the following:

The SIP method OPTIONS allows a UA to query another UA or a proxy
   server as to its capabilities.  This allows a client to discover
   information about the supported methods, content types, extensions,
   codecs, etc. without "ringing" the other party.  For example, before
   a client inserts a Require header field into an INVITE listing an
   option that it is not certain the destination UAS supports, the
   client can query the destination UAS with an OPTIONS to see if this
   option is returned in a Supported header field.  All UAs MUST support
   the OPTIONS method.
--

If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.

So it looks like your ITSP hasn't designed their system based on this RFC. When you send an OPTIONs message, the response has to include the capabilities the other party can support..

[Oleh Doroshenko]

Read the same before opening this topic, but still had hesitations. Ty for reply.