Hi guys,
We've been experiencing an intermittent one-way voice issue within our network where an external caller is unable to hear the internal called party.
The issue has been traced back to the CUBE that connects to our PSTN ITSP sending an invite to CUCM with a malformed RTP port that is neither within the range of RTP ports supported by the CUBE (16384-32766) or even a valid RTP port (m=audio 269101) Consequently, one-way audio is experienced.
SDL traces from CUCM also confirms that RTP port received in the SIP INVITE is malformed (below)
Sent:
INVITE sip:+XXXXXXXXX@172.16.81.75:5060 SIP/2.0
Via: SIP/2.0/TCP 172.16.82.16:5060;branch=z9hG4bK18CAD1205
From: <sip:XXXXXXXXX@ABC.org>;tag=1C2D81FA-2BB
To: <sip:+XXXXXXXXX@172.16.81.75>
Date: Sun, 09 Feb 2020 23:36:17 GMT
Call-ID: CDB602AD-4ACB11EA-AE23E289-897CF28@172.16.82.16
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3451237529-1254822378-2921194121-0144166696
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M7
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1581291377
Contact: <sip:XXXXXXXXX@172.16.82.16:5060;transport=tcp>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 28
P-Asserted-Identity: <sip:XXXXXXXXX@172.16.82.16>
Session-ID: a5dbc4b6fe145b5096654b4cdc13424c;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 318
v=0
o=CiscoSystemsSIP-GW-UserAgent 7680 6747 IN IP4 172.16.82.16
s=SIP Call
c=IN IP4 172.16.82.16
t=0 0
m=audio 269101 RTP/AVP 8 0 18 101
c=IN IP4 172.16.82.16
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20
The caller then places the call again immediately and the call would negotiate the RTP port fine and connect without any issues.
