Sep 1, 2007 - A Solution to the "10 Second Lost Audio Problem"

The Siemens Gigaset S450 IP/DECT phone is a product that promises much for anyone wanting both VoIP over SIP and PSTN connectivity with the clarity and convenience of a DECT cordless phone. The S450 delivers this and more, however users with the S450 and more basic C450/460 have found that they are not without their share problems and bugs, particularly when used with the Asterisk OpenSource PBX. For the S450, the so called 10 second lost audio problem is particularly irksome, but we have found a solution that works for us that we share here.

The lost audio problem typically occurs when calls are initiated via Asterisk to a remote party over SIP from an S450, and results in the first few seconds of audio from the called party being lost. According to some reports this can be as much as 10 seconds, although it was closer to 5 in our case. As soon as we hit the problem we felt that it should be resolvable and set to work with some analysis.

Enabling debugging on Asterisk 1.4.10.1, outgoing calls with the S450 showed a Packet2Packet Bridging step that was not present on external calls from other devices via Asterisk. Interestingly this step was also not shown when the S450 made internal VoIP calls, and in these cases the phones worked fine with no lost audio. The term "Bridging" immediately suggested the potential for RTP problems, and it did appear to be the cause.

We focussed on trying to disable the Packet2Packet bridging via sip.conf configuration, and after some time we found that setting dtmfmode to inband in sip.conf, and with the phone configuration using audio for the DTMF transport did the trick. Other options that we thought would be relevant such as canreinvite seemingly had no effect. We will be delving further into this issue with Asterisk, but if you have the problem currently, then try this setting and your problems may be resolved. Good luck!