[asterisk-bugs] [JIRA] (ASTERISK-28051) RTP engine should only accept audio frames with allowed payloads

[Jan Blom (JIRA)]

Jan Blom created ASTERISK-28051:
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             Summary: RTP engine should only accept audio frames with allowed payloads 
                 Key: ASTERISK-28051
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28051
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_rtp_asterisk
    Affects Versions: 15.6.0
         Environment: CentOS 7.4 with Asterisk 15.6.0, PJSIP channel only
            Reporter: Jan Blom


We run a stripped-down asterisk (both 15.5.0 and 15.6.0) with pjsip channel and only a few codecs since we want to avoid transcoding and other possible overhead.

Some incoming calls from one provider is setup with only PCMA in SDP from both sides. This usually works as expected. However, a few calls start with a single RTP packet with G.722 payload before we receive the G.711 stream.

This confuses Asterisk to think the received audio stream is G.722. “core show channel” reports “ReadFormat: g722” even after a second or two of receiving proper G.711 packets. 

Since we don’t have a translation path between our voice prompts and G.722, asterisk complains and call will eventually end with a failure. We don’t have the G.722 codec loaded.

Since we in the pjsip configuration only allow a few select codecs that we can handle, I would expect audio frames with a different payload  type to be ignored. At least as a configuration option.

Looking at the source, it seems that ast_rtp_codecs_get_payload() (main/rtp_engine.c) should return NULL if the received payload type is not found in the current rtp instance. I have attached a patch that solves the issue for us. 

There are probably other consequences with my simple patch, that I have overlooked. But it proves the point. 




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