sdp-anat on CME with 9971 SIP

[tato386]

I have several 9971 phones running SIP and working well on a CME 8.6 system. The CME also has plenty of SCCP phones running on it. I recently tried to add a 9971 phone that connects to the CME via VPN so therefore it is coming from a non-local subnet.

There is no firewall and no NAT involved. The phone registers but cannot call to any extension.

Doing a debug I basically see this: INVITE from 9971 require: sdp-anat from CME 420 bad extension unsupported: sdp-anat Why does the phone think it needs NAT? How do I tell the phone to not require this, or conversely, enable it on the CME to make the phone happy?

Thanks, Diego

[Mohammed al Baqari]

Hi,

ANAT is required for Dual mode support (IPv4/IPv6). Just make sure the preference configured in sip-ua is IPv4 only.

[tato386]

Mohammed,

That is a good point but I don't think it applies to my situation.  I don't have IPV6 enabled anywhere in my environment.  Also, I believe the sip-ua command would be implemented at the CME but it is the phone that is requesting the feature.  How would I disable the phone from requiring sdp-anat?

Thanks,

Diego

[Mohammed al Baqari]

I understand by the CME needs to be enabled for ANAT to accept the message from the phone. If the phone has ANAT in supported or required headers while CME isn't enabled for ANAT, CME will send 420 message. Just enable ANAT in your CME with preference mode IPv4 and test.

sip-ua

protocol mode dual-stack preference ipv4

[tato386]

Hello Mohammed,

I added the commands to the sip-ua and reloaded the CME for good measure.  It seems like IPV4/6 is now enabled but the phone is still getting the "420 bad extension" and "unsupported:sdp-anat" from the CME.  Here is what the sip-ua looks like:

UC520#sho sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED  192.168.1.100
SIP User Agent bind status(media): ENABLED  192.168.1.100
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is dual-stack, preference is ipv4
SDP application configuration:
 Version line (v=) required
 Owner line (o=) required
 Timespec line (t=) required
 Media supported: audio video image
 Network types supported: IN
 Address types supported: IP4 IP6
 Transport types supported: RTP/AVP udptl

[Mohammed al Baqari]

Ok. Can you please share the following.

sh run | sec voice register

debug ccsip messs ---- for a test call.

[tato386]

Here you are sir.  Sorry about the formatting of the debug.  I lost the line breaks and had to add them in by hand.  Thank you for all your help.

UC520#sh run | sec voice register
voice register global
 mode cme
 source-address 192.168.1.100 port 5060
 max-dn 160
 max-pool 40
 load 9971 sip9971.9-4-2SR2-2
 authenticate register
 authenticate realm all
 timezone 13
 voicemail 800
 tftp-path flash:
 file text
 create profile sync 7046320445592059
 camera
 video
voice register dn  1
 number 258
 name Conf Room
voice register dn  2
 number 259
 name ACA
voice register dn  3
 number 260
 name Extension 260
voice register pool  1
 id mac 3820.5618.5E20
 type 9971
 number 1 dn 1
 username 258 password *****
 description Conf Room
 codec g711ulaw
 no vad
 camera
 video
voice register pool  2
 id mac 38ED.18E9.6B15
 type 9971
 number 1 dn 2
 username 259 password *****
 description ACA
 codec g711ulaw
 no vad
 camera
 video
voice register pool  3
 id mac AC7E.8A2A.93AC
 type 9971
 number 1 dn 3
 username 260 password *****
 description Extension 260
 codec g711ulaw
 no vad
 camera
 video
UC520#

Jul 22 20:06:07.234: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:2@192.168.1.100;
user=phone SIP/2.0Via: SIP/2.0/UDP 10.100.20.11:5060;
branch=z9hG4bK2138dfbbFrom: "Extension 260" <sip:260@192.168.1.100>;
tag=ac7e8a2a93ac00150cb714be-6d8b8672To: <sip:2@192.168.1.100>Call-ID: ac7e8a2a-93ac000e-5013cc30-46176608@10.100.20.11Max-Forwards: 70Date: Fri, 22 Jul 2016 20:06:06 GMTCSeq: 101 INVITEUser-Agent: Cisco-CP9971/9.4.2Contact: <sip:260@10.100.20.11:5060;
transport=udp>;
videoExpires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Extension 260" <sip:260@192.168.1.100>;
party=calling;
id-type=subscriber;
privacy=off;
screen=yes
Require: sdp-anat
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.0.1Allow-Events: kpml,dialogRecv-Info: conferenceRecv-Info: x-cisco-conferenceAuthorization: Digest username="260",realm="all",uri="sip:2@192.168.1.100;
user=phone",response="631a01b9e50afd4f27a364cb73012692",nonce="98AB7E09005B356A",cnonce="1f8f4221",qop=auth,nc=00000003,algorithm=MD5Content-Length: 629Content-Type: application/sdpContent-Disposition: session;
handling=optionalv=0o=Cisco-SIPUA 17785 0 IN IP4 10.100.20.11s=SIP Callt=0 0m=audio 38590 RTP/AVP 0 102 9 124 8 116 18 101c=IN IP4 10.100.20.11a=rtpmap:0 PCMU/8000a=rtpmap:102 L16/16000a=rtpmap:9 G722/8000a=rtpmap:124 ISAC/16000a=rtpmap:8 PCMA/8000a=rtpmap:116 iLBC/8000a=fmtp:116 mode=20a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecvm=video 15634 RTP/AVP 97c=IN IP4 10.100.20.11b=TIAS:1000000a=rtpmap:97 H264/90000a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1a=imageattr:* recv [x=640,y=480,q=0.50]a=sendrecv

Jul 22 20:06:07.246: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 420 Bad Extension
Via: SIP/2.0/UDP 10.100.20.11:5060;
branch=z9hG4bK2138dfbbFrom: "Extension 260" <sip:260@192.168.1.100>;
tag=ac7e8a2a93ac00150cb714be-6d8b8672To: <sip:2@192.168.1.100>;
tag=3A8583C-45Date: Fri, 22 Jul 2016 20:06:07 GMTCall-ID: ac7e8a2a-93ac000e-5013cc30-46176608@10.100.20.11CSeq: 101 INVITEAllow-Events: telephone-event
Unsupported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.xContent-Length: 0

[Mohammed al Baqari]

Can you try this and see if it fixes the problem:

voice service voip

sip

no anat

voice register glo

cre pro

restart

[tato386]

Hello Mohammed,

Unfortunately I am still seeing "required: sdp-anat" messages from the phone followed by "420 bad extension" and "unsupported: sdp-anat" from the CME.

I would think there would be a way to modify the phone so that it does require sdp-anat but have not found an option to do this.

Do you think we would have more luck if we concentrate on the phone itself?

Rgds,

Diego

[Mohammed al Baqari]

Hi Mate,

The response is coming from the phone which indicates that the feature is enabled on the phone.