traud / asterisk-gsm-efr Goto Github PK

When Asterisk started the call but the remote party sends a re-INVITE on the SIP layer, the call might face one-way audio. One such a call scenario is call hold, for example. One-way audio happens because this codec here uses a dynamic RTP payload type in SDP negotiation, which Asterisk 13 LTS does not support correctly. Complete details are documented in ASTERISK-27056. Even if you are not affected, applying the patch provided there is recommended for any user. If you face an issues with that patch, report in ASTERISK-27056, please. If possible, consider upgrading to a newer Asterisk branch.

Currently, efrtolin_framein expects only a single frame per RTP payload. Therefore, when the ptime is higher than 20ms, audio quality is going to degrade. Actually, just a for-loop is missing, very much like the one in my Codec 2 module. However, Google Android does not send more than one frame per payload. Therefore, I do not have a working GSM-EFR client to test. If you need this feature, please, reply here and mention your VoIP/SIP client.

• Nokia Series 40, like Nokia X3-02, and
• Asha Software Platform, like Nokia Asha 501

support GSM-EFR. By the way, Symbian/S60 does not offer GSM-EFR. However, the audio sounds distorted in both directions. sRTP did not make a difference. GSM-EFR not transcoded but passed-through works between those clients. Pass-through with the Google Android implementation gives the same distorted audio. Google re-orders the speech bits like in AMR. Perhaps, Nokia does not do this re-ordering. That should be tested first.

All these clients support AMR.