06-16-2011 09:27 AM - edited 03-19-2019 03:07 AM
Hi all. I am practicing voice on my dynamips router using the following IOS
C3725-ADVIPSERVICESK9-M), Version 12.4(15)T6
Now i am using 2 PCs with xlite phones. Following is my routers relevant config
voice register global
mode cme
source-address 1.1.1.1 port 5060
max-dn 10
max-pool 2
authenticate register
timezone 23
time-format 24
date-format D/M/Y
create profile sync 0000837224505145
voice register dn 1
number 3003
voice register dn 2
number 3004
voice register pool 1
id mac 001E.EC7A.435A
number 1 dn 1
dtmf-relay rtp-nte
username 3003 password 3003
description 0207033003
codec g711alaw
voice register pool 2
id mac 0000.0000.0000
number 1 dn 2
voice service voip
allow-connections sip to sip
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server expires max 120 min 60
Now both both phones are registering and able to make call to each other. When under pool 1 and pool 2, i change the codec to g729 (default), then i get media not acceptable error !!. Here is the debug
*******************************************************************************
R1#deb ccsip messages
SIP Call messages tracing is enabled
R1#
*Mar 1 00:12:31.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:3004@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:45950;branch=z9hG4bK-d87543-25091602d57eb162-1--d
87543-;rport
Max-Forwards: 70
Contact: <sip:3003@192.168.2.100:45950>
To: "3004"<sip:3004@1.1.1.1>
From: "3003"<sip:3003@1.1.1.1>;tag=dd0dbc1c
Call-ID: eb41e96ea832256cMmQ3ZjU0YzAwZjFlYjliM2ZkMTZlNGYwYWU1NmU5OGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF
O
Content-Type: application/sdp
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 384
v=0
o=- 8 2 IN IP4 192.168.2.100
s=<CounterPath eyeBeam 1.5>
c=IN IP4 192.168.2.100
t=0 0
m=audio 23192 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : T4mNNpnc nQiE56kE 192.168.2.100 23192
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:9F4E771B281
R1#948939404E8EAB27B1E70
*Mar 1 00:12:31.459: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.2.100:45950;branch=z9hG4bK-d87543-25091602d57eb162-1--d
87543-;rport
From: "3003"<sip:3003@1.1.1.1>;tag=dd0dbc1c
To: "3004"<sip:3004@1.1.1.1>;tag=B7740-698
Date: Fri, 01 Mar 2002 00:12:31 GMT
Call-ID: eb41e96ea832256cMmQ3ZjU0YzAwZjFlYjliM2ZkMTZlNGYwYWU1NmU5OGU.
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=65
Content-Length: 0
*Mar 1 00:12:31.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:3004@1.1.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:45950;branch=z9hG4bK-d87543-25091602d57eb162-1--d
87543-;rport
To: "3004"<sip:3004@1.1.1.1>;tag=B7740-698
From: "3003"<sip:3003@1.1.1.1>;tag=dd0dbc1c
Call-ID: eb41e96ea832256cMmQ3ZjU0YzAwZjFlYjliM2ZkMTZlNGYwYWU1NmU5OGU.
CSeq: 1 ACK
Content-Length: 0
**************************************************************************
Since i am using dynamips, is it related to DSP issues ?
06-16-2011 11:20 AM
This is because your voice register pool is set to require codec g711alaw. This is the only codec that CME will accept. The INVITE from the xlite phone is not offering that. You can create a voice class codec and assign that to the pool so CME will accept more than one codec.
PS- This is the line that offers G.729:
a=rtpmap:18 G729/8000
This is what G711 would look like:
a=rtpmap:0 pcmu/8000 or
a=rtpmap:0 pcma/8000
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