IP Troubleshooting

Properly troubleshooting the IP call leg is critical for fax, modem, and text telephony calls. Because these modulated communications were designed strictly for the PSTN, transporting them over IP adds additional challenges. Making sure the IP portion of the call path is operating as reliably and efficiently as possible is necessary for ensuring that fax, modem, and text calls are successful.

Although the previously discussed show call active voice brief command is helpful for detecting problems on the IP call leg for IOS gateways, additional tools are available for more detailed troubleshooting. On the Cisco voice gateways, additional commands provide statistical views of the IP packets received from the DSP perspective and packet captures with third-party tools provide the ultimate look at the IP call leg.

Cisco voice gateways provide views of the incoming IP media stream from the digital signal processor (DSP) perspective. This IP media stream carries the packetized fax or modem information. The DSP on the voice gateway can track information about the media stream that enables you to determine whether network problems are having a detrimental effect on the fax/modem/text call.

Within the DSP, a playout or jitter buffer is responsible for handling the individual packets that make up the incoming media stream. This playout buffer defines a fixed amount of delay for passthrough and relay calls to compensate for IP network impairments such as delay, jitter, and out-of-sequence packets. Figure 12-6 illustrates the concept of a playout buffer and how it handles the incoming IP packets.

Figure 12-6 DSP Playout Buffer on an IOS Gateway for a T.38 Fax Relay Call

Variably spaced

T.38 packets arrive at the jitter buffer and some may even be out of sequence

Packets are re-sequenced if necessary and placed in the correct order for playout

Evenly spaced packets are fed to the DSP CODEC for transmission to the PSTN

Evenly spaced packets are fed to the DSP CODEC for transmission to the PSTN

300 ms Fixed Playout Buffer (Default setting for fax relay)

In Figure 12-6, you can see a media stream of T.38 fax relay packets entering the playout buffer from the IP network. For fax relay, the playout buffer is set to a fixed value of 300 ms in Cisco gateways. Sequence numbers carried by the T.38 UDPTL encapsulation are used to resequence the packets and line them up in the correct order for playout by the DSP codec to the PSTN. In the case of passthrough and Cisco fax relay, RTP sequence numbers are used rather than T.38 UDPTL.

For passthrough, the playout buffer is also set to a fixed value, but this value may vary. In IOS gateways, the fixed playout buffer size for modem passthrough and fax pass-through calls is actually derived from the last playout buffer setting in use by the voice call before the switchover occurs. Unlike the fixed buffer sizes for modem and fax calls, voice calls use an adaptive playout buffer, and a wide range of values are possible. When the call transitions to passthrough, the current adaptive voice playout buffer setting is converted into a fixed buffer size.

When dealing with long modem or fax passthrough calls, be cognizant of the lack of DSP clock synchronization that may cause these longer calls to fail. This issue was discussed in detail in the section "Timing and Synchronization" in Chapter 7, "Design Guide for Fax,

Modem, and Text." If you run into this issue, it is recommended that a relay transport method, such as T.38, Cisco fax relay, or Cisco modem relay, be used because relay transport methods do not suffer from this problem.

Troubleshooting the IP leg of a fax, modem, or text call can vary depending on whether you have an IOS gateway or a non-IOS gateway. For this reason, IOS and non-IOS gateways are addressed in separate sections. Packet captures can also be used in IP troubleshooting regardless of the gateway type, and these are discussed in the last section.

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