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Cisco Bug: CSCvs94318 - SIP-GW is not complaint to v.150.1 MER

Last Modified

Mar 24, 2020

Products (1)

  • Cisco 3900 Series Integrated Services Routers

Known Affected Releases

15.5(1.1.49)SY4

Description (partial)

Symptom:
Customer trying to set up below call flow for secure modem relay.Its a defense company and they are trying to implement v150.1 MER complaint modem calls

STE > FXS > VG310 SCCP > SCCP > CUCM > SIP > SIP-GW (3925) > T1 PRI > STE

After the call connect,VG310 SCCP GW will receive ORC/SMT from CUCM but it was missing MER capabilities .Post this call connected but call proceeded as modem pass-through call instead of modem relay call.Worked with CUCM and GW BU's with BEMS01052730(GW) and BEMS01053193 (CUCM) and they confirmed below incapability of SIP-GW which is causing this behavior.Repose from CUCM BU

?The issue is due to the absence of the  parameter mrmods which specifies the V.34 duplex and V.32bis in the 183 session progress and in the 200 OK.In the CUCM condition we check if the modulations have the V.34 duplex, V.32bis and the RFC2833 events (32-35).As the mrmods parameter is not present the check fails. The same behaviour is seen in the previous and the new logs"

## The validation fails with reason -4 (MISSING_MODS_OR_ANS_EVENTS)
 
70670736.010 |22:30:34.871 |AppInfo  |//SIP/SIPHandler/ccbId=0/scbId=0/getV150Attributes: Incomplete or non-compliant MER offer detected. Validation Code:-4
 
Code Snippet:
 
// Next we check modulations (V.34 duplex and V.32bis must be supported) and
    // RFC2833 events (all four answer events must be supported).
    //
    ModulationAndModem2833Support expectedMod2833Caps =
        V150_MOD_V34_DUPLEX | V150_MOD_V32_V32BIS | V150_RFC2833_ANS | V150_RFC2833_ANSAM | V150_RFC2833_ANS_RVS_PHASE | V150_RFC2833_ANSAM_RVS_PHASE;
 
    if (!v150Caps.areTheseModemRelayModAndToneEventsSet(expectedMod2833Caps))
    {
        return MISSING_MODS_OR_ANS_EVENTS;
    }

Outbound INVITE from CUCM to SIP-GW with "mrmods=1-12" on the last attribute line
 
INVITE sip:14084000875@172.28.164.129:5060 SIP/2.0
Via: SIP/2.0/TCP 172.23.105.218:5060;branch=z9hG4bK1f05a59fbcab1
From: <sip:4087433465@172.23.105.218>;tag=41151469~8472f2f5-a7bd-42ef-a8c7-783a4ff018d8-44613694
To: <sip:14084000875@172.28.164.129>
Date: Fri, 31 Jan 2020 03:30:33 GMT
Call-ID: 84f1800-e3319f59-1e9a2b-da6917ac@172.23.105.218
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM11.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.23.105.218:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Session-ID: 9458856fc08ad64982e222a1d798bba0;remote=00000000000000000000000000000000
Cisco-Guid: 0139401216-0000065536-0000004032-3664320428
Session-Expires:  1800
P-Asserted-Identity: <sip:4087433465@172.23.105.218>
Remote-Party-ID: <sip:4087433465@172.23.105.218>;party=calling;screen=yes;privacy=off
Contact: <sip:4087433465@172.23.105.218:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 627
 
v=0
o=CiscoSystemsCCM-SIP 41151469 1 IN IP4 172.23.105.218
s=SIP Call
c=IN IP4 172.28.164.122
b=TIAS:64000
b=AS:64
t=0 0
m=audio 16426 RTP/AVP 0 8 116 18 101 118 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:120
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-35
a=rtpmap:118 v150fw/8000
a=rtpmap:126 NoAudio/8000
a=sqn:0
a=cdsc: 1 audio RTP/AVP 0 8 116 18 101 118 126
a=cdsc: 8 audio udpsprt 120
a=cpar: a=sprtmap:120 v150mr/8000
a=cpar: a=fmtp:120 mr=1;mg=0;CDSCselect=1;jmdelay=no;versn=1.1;mrmods=1-12
a=vndpar:2 9 2 5
 
SIP-GW response.?a? attribute is missing mrmods=1-12
 
## Incoming 183 Session Progress without the parameter mrmods
 
70670735.002 |22:30:34.870 |AppInfo  |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 172.28.164.129 on port 5060 index 3 with 1354 bytes:
[110703938,NET]
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 172.23.105.218:5060;branch=z9hG4bK1f05a59fbcab1
From: <sip:4087433465@172.23.105.218>;tag=41151469~8472f2f5-a7bd-42ef-a8c7-783a4ff018d8-44613694
To: <sip:14084000875@172.28.164.129>;tag=B5EBDFA8-155F
Date: Fri, 31 Jan 2020 03:30:33 GMT
Call-ID: 84f1800-e3319f59-1e9a2b-da6917ac@172.23.105.218
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:4084000875@172.28.164.129>;party=called;screen=no;privacy=off
Contact: <sip:14084000875@172.28.164.129:5060;transport=tcp>
Supported: sdp-anat
Supported: X-cisco-srtp-fallback
Server: Cisco-SIPGateway/IOS-15.6.3.M6a
Session-ID: b7cebd9648ce5037a649cc35d694ca96;remote=9458856fc08ad64982e222a1d798bba0
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 430
 
v=0
o=CiscoSystemsSIP-GW-UserAgent 3058 3024 IN IP4 172.28.164.129
s=SIP Call
c=IN IP4 172.28.164.129
t=0 0
m=audio 25508 RTP/AVP 0 101 118
c=IN IP4 172.28.164.129
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,32-35
a=rtpmap:118 v150fw/8000
a=fmtp:118 1,3
a=sqn:0
a=cdsc: 1 audio udpsprt 120
a=cpar: a=sprtmap:120 v150mr/8000
a=cpar: a=fmtp:120 mr=1;mg=0;CDSCselect=1;jmdelay=no;versn=1.1

Because of this,CUCM fails to verify the V.150 MER capabilities of SIP-GW and instruct VG310 SCCP GW with normal modem pass-through call.

This was informed to GW BU and they confirmed that at present SIP-GW doesn't have the capability for v150.1 MER and need to file a defect to add that attribute

GW modem : 3925
IOS : 15.6.3.M6a
GW BU response

"The CUCM Team's analysis makes sense. SIP GW support for V.150.1 implementation is  not compliant to the V.150.1 MER requirements.Please engage the CTG SIP team by opening a defect against the voice-sip component with the details that you have shared here. They should be able to address this attribute settings. Please remember that the SIP GW supports only V.32 and V.34 modulation as far as MR support goes...so mrmods will not actually be 1-12."

Conditions:
v150.1 MER calls passing through SIP-GW <> PRI
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