RTP stream one way in CME with SIP trunk from service provider.

[abdullah alnahdi]

Dear Expert

I am facing an issue for outgoing and incoming calls. The issue is the RTP is only one way the ip phone can hear the voice for incoming and outgoing calls and PSTN not hearing.. i did some debug as per attachment for incoming and outgoing calls. 

debug ccsip message

debug voip ccapi inout

Call details:

INCOMING calls---calling number is 0555185650---called number 009665104400-----received only last 4 digit from service provider 4400--translated to 1000 (internal extension) by num-exp---phone ring but PSTN not hear only ip phone hear.

OUTGOING calls----calling number 009665104400 from extension 1000----called number is 00966598621604----Cisco ip phone hear only

Your support is highly appreciate

[nathanjgrace · Accepted solution]

Hi Abdullah, I'm curious about the interfaces to which your media is bound.  It looks like your SIP media is bound to g0/0.5, which is 10.10.5.1/24, however, all your dial peers are bound to g0/1, which is 192.168.2.2/30.

The carrier SBC looks like it's defined as 10.128.12.132, and the route is going through 192.168.2.1. Is it possible that 192.168.2.1 doesn't have the correct route for the 10.10.5.0 network?

Finally, I see in the INVITE that media is being sent from 10.128.12.133 (c=IN IP4 10.128.12.133) from the carrier side.  Do you have routing for this set up or is the firewall blocking it?  Perhaps add the below command to your gateway:

ip route 10.128.12.133 255.255.255.255 192.168.2.1

I hope to hear the results!

[abdullah alnahdi]

Dear 

Thank you very much for your help. I really appreciate that.

There is no firewall. It is a sample configuration which is Voice gateway acts as CME and connected to switch then ip phones are connected to the switch.

The ip route is there already as per show run 

any other solution.

[Manish Gogna]

Can you plz remove the following from the dial-peers and check

voice-class sip bind control source-interface GigabitEthernet0/1
 voice-class sip bind media source-interface GigabitEthernet0/1

Manish

[abdullah alnahdi]

Dear Manish 

Should i remove from both incoming and outgoing dial-peer.

[Manish Gogna]

Hi Abdullah,

Since you have defined a diff interface under voice service voip i would suggest removing it from all dial-peers as they would take preference over the ones defined under voice service voip.

Manish

[abdullah alnahdi]

Friend

Thanks for your suggestion i will check and update you.

[nathanjgrace]

You have the route for 10.128.12.132 in your config.  The carrier appears to be using .132 for signaling but .133 for media.  Check the SDP section of the INVITE:

*Jul 31 14:48:52.911: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received: 
INVITE sip:4400@192.168.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.128.12.132:5060;branch=z9hG4bK83601D52
Remote-Party-ID: <sip:0555185650@10.128.12.132>;party=calling;screen=no;privacy=off
From: <sip:0555185650@10.128.12.132>;tag=1BB18B8-17AD
To: <sip:4400@192.168.2.2>
Date: Sun, 31 Jul 2016 14:39:42 GMT
Call-ID: 73E99C34-566311E6-8F438DB4-E1531967@10.128.12.132
Supported: 6,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1944609611-1449333222-2403175860-3780319591
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1469975982
Contact: <sip:0555185650@10.128.12.132:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 248

v=0
o=CiscoSystemsSIP-GW-UserAgent 6752 53 IN IP4 10.128.12.132
s=SIP Call
c=IN IP4 10.128.12.133
t=0 0
m=audio 16494 RTP/AVP 8 101
c=IN IP4 10.128.12.133
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

[abdullah alnahdi]

Dear 

Thanks a lot for helping and support.Now we can hear each other but the call dropped after sometimes (80 sec) i attached the debug.