TD 92685EN
28 June 2012 / Ver. A
Troubleshooting Guide
Ascom i62 VoWiFi Handset
46
6. Troubleshooting
The PBX updates the 180 RINGING with the name "Freddy" in the P-Preferred-Identity, thus
asserting “Freddy” as the name of the called party. The 180 RINGING is sent back to the
originator of the call so that “Freddy” is presented to the originator before the call is
actually answered:
180 Ringing
From: "9910" <sip:9910@10.11.24.244>;epid=00013e124af6;
tag=2363920757
To: "Freddy" <sip:9920@10.11.24.244;user=phone>;
tag=3629151891
Call ID: ab5e5a3de909d311b77400013e124af6@10.11.24.177
Contact: <sip:9920@10.11.24.244:5060;user=phone;transport=UDP>
P-Preferred-Identity: "Freddy" <sip:9920@10.11.24.244;user=phone>
The final ACKS and 200 OK result in the RTP
medi
a session being established.
6.1.3 Identifying Problems
A problem that can occur is that one party cannot hear the other party. I
n this example
scenario, an analysis of the trace reveals that despite the problem, signaling has been
completed successfully. In this situation the support engineer should:
• Take a Wireshark trace and expand the message headers to check f
or errors in the SDP
negotiation, such as codecs or ports. To do this, the handset needs to be setup to support
RPCAP logging as described in Appendix D.2.
• Take a trace from the handset that is unable to h
ear the other party and
establish
whether or not RTPs packets are being sent to the handset.
• Determine where RTPs are being sent, for example, are they being sent to th
e correct
port or the correct IP address.
Check SDP Negotiations
There are a number of different SDP negotiations
between endpoints that can be considered
and investigated in response to problems such as one way audio. The negotiations that
should be focused on and investigated through a trace from the wired LAN are:
• Port negotiation. Check that the correct port number
s are used for the RTP session.
• IP address negotiation: Check whether or not the other party has up
dated its SDP with its
IP address.
• Payload negotiation: Whether the co
rrect codec has been configured.
• DTMF negotiation. DTMF may be generated inband or through SIP info.
• Packet interval negotiation: check whether the RTP packet interval b
etween the
endpoints is too great. For example, one way audio may be the result of the receiving
end supporting a minimum packet interval of 30 ms while the sending end negotiates
that packets are sent every 20 ms.
Send the trace to Ascom for further analysis.
Make sure that the attributes of the SDP signaling on either side are correct and t
hat the
RTP stream is bidirectional.
Check that one of the parties is not still on hold.
Refer to RFC 3665 for additional information. This descri
bes the best common practice for
SIP calls.
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