Hi, when testing the integrated SIP Client on the Nexus S (Android 2.3.4) against FRITZ!Box CPEs (highly spread WiFi/VoIP router across Europe) we experience the following issue:
When placing an outgoing or incoming VoIP call over the CPE the playback of the incoming RTP audio is scrambled. The outgoing RTP audio is alright. The reason for that is that the CPE uses 30 ms packetization for the G. 711 data. The audio playback works alright when the packetization is adjusted to 20 ms. As it is highly common here to have 30 ms packetization for G.711 we badly need 30 ms support in the native SIP stack in android. Is there any chance to get that support into the framework? Usually that can be easely done by only adjusting the buffer size and allowing 30 ms packets to be received. For the corresponding CPE it would be alright if received 30 ms packets would be processed and still packets would be sent using 20 ms packetization. Thanks Sven -- You received this message because you are subscribed to the Google Groups "Android Developers" group. To post to this group, send email to [email protected] To unsubscribe from this group, send email to [email protected] For more options, visit this group at http://groups.google.com/group/android-developers?hl=en
