Change Control:

Versi
on:

Dat
e:

1.0

6th
Aug

Chan
ged
by:
Awais

Description

Draft document for task 2.6e

TASK 2.4E: HQ SIP TRUNK REDUNDANCY O PSTN

STEPS:
CUCM:

We have to use the previously created SIP Trunk to CUBE for Intersite
calling using ILBC Codec
Go to the Route list created in task 2.3b
Add the CUBE-ILBC trunk in the Route list
Adjust the Calling ID transformation mask to be 8202XXXX

CUBE:
Create a dial-peer towards PSTN
dial-peer voice 15 voip
destination-pattern 85151111
session protocol sipv2
session target ipv4:157.26.1.253
incoming called-number 85151111
codec transparrent
dtmf-relay rtp-nte
no vad
Make a call from HQ Phone to PSTN Phone, call should be established, most
probally black screen will come on PSTN Phone
Enable the following debug:
Debug ccsip messages

Check the message in which IP address of PSTN appeared in VIA

Header
See media capabilities by finding the below message in which you see
atleaset following:

c=IN IP4 157.26.1.

SAMPLE MESSAGE:
ul 15 10:37:35.370: //212/661200800000/SIP/Msg/ccsipDisplayMsg:

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 142.102.64.254:5060;branch=z9hG4bK12D1215
From: "HQ Phone 2" <sip:82022002@142.102.64.254>;tag=92ABC0-BDD
To: <sip:85151111@157.26.1.253>;tag=2977CC28-1E2A
Date: Tue, 15 Jul 2014 10:37:35 GMT
Call-ID: D2D7D597-B4211E4-80E1CE0F-E62CD3AF@142.102.64.254
Timestamp: 1405420636
CSeq: 104 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Video PSTN Phone"
<sip:85151111@157.26.1.253>;party=called;screen=yes;privacy=off
Contact: <sip:85151111@157.26.1.253:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M6a
**This show it is CME on
PSTN ***
Supported: timer
Content-Type: application/sdp
Content-Length: 328
v=0
o=CiscoSystemsSIP-GW-UserAgent 3270 1973 IN IP4 157.26.1.253 **This
show it is CME on PSTN ***
s=SIP Call
c=IN IP4 157.26.1.253
t=0 0
m=audio 17490 RTP/AVP 18 101
c=IN IP4 157.26.1.253
a=inactive
a=rtpmap:18 G729/8000 *** It shows PSTN is using G729 ***
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
m=video 0 RTP/AVP 119
*** It shows RTP Payload PSTN is using, in
this case 119 is being used ***
c=IN IP4 157.26.1.253

From above message, we observe the following:

- PSTN side is CUCME
- Codec g729 is being used

RTP Payload 119 is being used

We have to adjust the dial-peer accordingly,

dial-peer voice 15 voip

destination-pattern 85151111
session protocol sipv2
session target ipv4:157.26.1.253
incoming called-number 85151111
rtp payload-type cisco-codec-fax-ack 111
rtp payload-type cisco-codec-video-h264 119
codec transparrent
dtmf-relay rtp-nte
no vad