Sip Trunk - No audio - STREAM_DEAD

[Martin Abildgaard]

Hi all

Have a little issue with a SIP-trunk from Optimum connected via CUBE to a Cisco CUCM 8.6.2

Callflow: Provider SIP -> CUBE -> SIP -> CUCM -> SCCP phone

When inbound call is answered on the phone there is no audio either way. The odd part is when I do Show sip call the call leg between the provider and CUBE the status is STREAM_DEAD

Outbound call no issues.

Attached is debug, status and config

Any idea why?

The trunk has been working, until I was testing some DTMF from a 9951.. but cannot find out what is the issue..

[Suresh Subramanian]

Hello Martin,

when the inbound call is answered by phone, the CUCM sends 200 OK to CUBE.

*Jan 24 21:49:36.523: //38/3BE4116F8047/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.102.61.10:5060;branch=z9hG4bK16192A

From: "Name Unavailabl" <00045XXXXXX>;tag=265B68-B37

To: <>;tag=42094~b6d41696-648e-4f33-875e-543830787a17-25000868

Date: Fri, 24 Jan 2014 21:59:00 GMT

Call-ID: 3BE4AD97-847811E3-804DC245-95567DD5@10.102.61.10

CSeq: 101 INVITE

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

Allow-Events: presence, kpml

Supported: replaces

Call-Info: <10.2.1.51:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires:  14400;refresher=uas

Require:  timer

Contact: <>

Content-Type: application/sdp

Content-Length: 233

v=0

o=CiscoSystemsCCM-SIP 42094 1 IN IP4 10.2.1.51

s=SIP Call

c=IN IP4 10.102.61.10 ---> IP address to receive the RTP Stream (Phone's IP address)

b=TIAS:64000

b=AS:64

t=0 0

m=audio 16422 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000 ---> G711 U-law codec

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

However when the CUBE sends back the 200 OK to ITSP, it replies with IP address 0.0.0.0

*Jan 24 21:49:36.527: //37/3BE4116F8047/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.10.100.1:5060;branch=z9hG4bK302578409

From: "Name Unavailabl" <45XXXXXX>;tag=B9x%AG6dn7LmjNB0I3731E9A37f49ed4

To: ;tag=265BE8-1236

Date: Fri, 24 Jan 2014 21:49:28 GMT

Call-ID: 2075305078@10.10.100.1

CSeq: 1138 INVITE

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

Allow-Events: telephone-event

Remote-Party-ID: <>;party=called;screen=no;privacy=off

Contact:

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.4.1.T

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 224

v=0

o=CiscoSystemsCCM-SIP 42094 1 IN IP4 10.2.1.51

s=SIP Call

c=IN IP4 0.0.0.0

b=TIAS:64000

b=AS:64

t=0 0

m=audio 0 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

>> so your CUBE is not responding to ITSP with correct IP address. it could be issue with media interface binding.

could you please post your cube config?

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[Martin Abildgaard]

good observation, thanks.

Have been through the config, but should be fine.. Could it be that the interface the outbound dialpeer is bound to is DHCP?

Attached is the config... forgot it in the first post..

[Suresh Subramanian]

Hello Martin,

Yes, you are right. There is no static ip configured in the interface bound towards provider.

interface GigabitEthernet0/1

ip address dhcp

could you please try configuring an ip address to this ip address ensure it has proper routing towards other interfaces?

you may refer the below URL on how media & signaling binding works.

CUBE SIP Media and Signalling Binding to an Interface

https://supportforums.cisco.com/community/netpro/collaboration-voice-video/ip-telephony/blog/2013/04/04/cube-sip-media-and-signalling-binding-to-an-interface

//Suresh

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