[cisco-voip] IOS and RFC 3556 support

[Ryan Ratliff (rratliff)]

Something else to keep in mind is that unless you are on the latest phone load (9.3.4, 9.4.1, or later) pretty much every Cisco SIP IP phone will fail a call that comes in with these parameters in the SDP.  

There's an option in the SIP Profile in UCM 10 that will allow you to filter these at the server.  I'd strongly recommend using RTCP to avoid these showing up to begin with.

-Ryan

On Nov 14, 2013, at 9:25 AM, Brian Meade (brmeade) <brmeade at cisco.com> wrote:

Also if this is incoming, this issue was resolved in CSCtb74536- http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtb74536

Brian

-----Original Message-----
From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Meade (brmeade)
Sent: Thursday, November 14, 2013 9:23 AM
To: Andreas Sikkema; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS and RFC 3556 support

Where do you have the SIP Profile applied?  Is that an Incoming 183 Session Progress?  Right now, CUBE can only modify outgoing messages with SIP-profiles.

Brian

-----Original Message-----
From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andreas Sikkema
Sent: Thursday, November 14, 2013 9:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS and RFC 3556 support

Hi,

We have a couple of Cisco IOS based VoIP machines (AS5350XM ISDN30 gateways and 2801 CUBEs) and when they receive a call containing specific RFC 3556 information in the SDP calls are rejected.

The only reference to RFC 3556 I can find on the Cisco site is related to the SBC functionality on an ASR1000
(http://www.cisco.com/en/US/docs/routers/asr1000/configuration/guide/sbcu/sbc_sdpbw.html)
but that doesn't mention the specific fields I am having problems with.

Here's an example of a message received by the Cisco machines containing the fields:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxxxxxxx:5060;branch=z9hG4bK4c35.5b48e4b.0
Via: SIP/2.0/UDP
xxxx:5060;received=xxxxx;rport=62955;x-route-tag="cid:blah";branch=z9hG4bK2D3293314D4
From: <sip:0xxxxxxxxx at xxxxxxxx>;tag=518D974C-198D
To: <sip:0xxxxxxx at xxxxxxx>;tag=SDcaa1899-227626126866920131113103043
Call-ID: xxxx at yyyyy
CSeq: 101 INVITE
Record-Route: <sip:xxxxxxxxx;lr;ftag=518D974C-198D;did=1f4.fdba577>
Require: 100rel
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
RSeq: 913
Contact: <sip:+31xxxxxxxx at xxxxxxxxxx:5060;transport=udp>
Supported: 100rel
Content-Type: application/sdp
Content-Length: 258

v=0
o=- 13549184 13549184 IN IP4 xxxxxxxx
s=-
c=IN IP4 xxxxxxxx
t=0 0
a=sendrecv
m=audio 49980 RTP/AVP 8 101
c=IN IP4 xxxxxxxxx
b=RR:0
b=RS:0
a=rtpmap:8 PCMA/8000
a=maxptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

The problem are the b=RR:0 and b=RS:0 fields.

I've tried removing them using a sip profile on the correct dial-peer:

voice class sip-profiles 200
request ANY sdp-header Audio-Bandwidth-Info remove  response ANY sdp-header Audio-Bandwidth-Info remove  request ANY sdp-header Video-Bandwidth-Info remove  response ANY sdp-header Video-Bandwidth-Info remove  request ANY sdp-header Bandwidth-Key remove  response ANY sdp-header Bandwidth-Key remove

I am seeing the profile being used when doing a test call and having debug enabled, but the behaviour doesn't change and one of the bandwidth lines is not being hit.

Has anyone seen this before? I can try to remove the fields elsewhere in our platform, but that would mean spending loads of CPU cycles in places where just a part of the calls really need this fix. I'd rather fix this at the source of the problem.

Has anyone seen this before and fixed it?

--
Andreas Sikkema
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip