v4.11.0

What's Changed

• core: cleanup stream_rtp_h ignore handling by @sreimers in #3770
• audio: increase RTP timestamp also for underruns by @cspiel1 in #3775
• aubuf: remove adaptive mode by @cspiel1 in #3780
• webrtc_aec: fix system cmake include by @sreimers in #3782
• call: avoid call progress if media is inactive by @cspiel1 in #3783
• conf: remove unused conf_aubuf_adaptive() by @alfredh in #3785
• audio: reuse mbuf for telephony events by @maximilianfridrich in #3787
• call: add callback for incoming SIP INFO by @SnowMaster12 in #3762
• test: check fixture error and fix call progress by @alfredh in #3786
• aureceiver: fix return on error case in aurecv_start_player() by @alfredh in #3792
• audio: fix warning() %d format in encode_rtp_send() by @alfredh in #3793
• opus,aureceiver: fix PLC/FEC handling by @sreimers in #3794
• l16: refactor NR_CODECS by RE_ARRAY_SIZE by @sreimers in #3803
• config: remove unused config option 'audio_silence' by @alfredh in #3798
• echo: refactor call_event_handler closed with bevent by @sreimers in #3806

Full Changelog: v4.10.0...v4.11.0