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fix: WebRTC codec mismatching #141

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Merged
shenjinti merged 1 commit into from
Feb 27, 2026
Merged

fix: WebRTC codec mismatching #141

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161 changes: 78 additions & 83 deletions src/proxy/proxy_call/session.rs
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Original file line number Diff line number Diff line change
Expand Up @@ -1308,42 +1308,42 @@ impl CallSession {
answer = answer_for_caller;

if self.media_bridge.is_none() {
// codec_a: use caller's OFFER to determine what
// they actually send (first codec in m= line)
let (params_a, dtmf_pt_a, codec_a) = {
let from_offer = self.caller_offer.as_ref().and_then(|offer| {
let (codecs, dtmf) =
MediaNegotiator::extract_codec_params(offer);
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()?;
Some((chosen.to_params(), dtmf, chosen.codec))
});
from_offer.unwrap_or_else(|| {
self.answer
.as_ref()
.map(|s| {
let (codecs, dtmf) =
MediaNegotiator::extract_codec_params(s);
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()
.unwrap_or(CodecInfo {
payload_type: 0,
codec: CodecType::PCMU,
clock_rate: 8000,
channels: 1,
});
(chosen.to_params(), dtmf, chosen.codec)
})
.unwrap_or((
rustrtc::RtpCodecParameters::default(),
None,
CodecType::PCMU,
))
})
let caller_is_webrtc = self
.caller_offer
.as_ref()
.map(|offer| Self::is_webrtc_sdp(offer))
.unwrap_or(false);
let from_offer = self.caller_offer.as_ref().and_then(|offer| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(offer);
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()?;
Some((chosen.to_params(), dtmf, chosen.codec))
});
let from_answer = self.answer.as_ref().map(|s| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(s);
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()
.unwrap_or(CodecInfo {
payload_type: 0,
codec: CodecType::PCMU,
clock_rate: 8000,
channels: 1,
});
(chosen.to_params(), dtmf, chosen.codec)
});
let default_codec = (
rustrtc::RtpCodecParameters::default(),
None,
CodecType::PCMU,
);
let (params_a, dtmf_pt_a, codec_a) = if caller_is_webrtc {
from_answer.or(from_offer).unwrap_or(default_codec)
} else {
from_offer.or(from_answer).unwrap_or(default_codec)
};

// codec_b: use callee's ORIGINAL early media SDP
Expand Down Expand Up @@ -1693,53 +1693,48 @@ impl CallSession {
CodecType::PCMU,
));

// For params_a (caller side), determine what codec the caller will
// actually send. The caller's OFFER m= line order reflects their
// sending preference. Using select_best_codec on the answer is wrong
// because the caller may ignore the answer's codec ordering and send
// whichever codec it prefers (e.g., active-call forwards carrier's
// PCMA regardless of our answer's codec preference).
let (params_a, dtmf_pt_a, codec_a) = {
// First try: use caller's offer (what they actually send)
let from_offer = self.caller_offer.as_ref().and_then(|offer| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(offer);
// First codec in offer's m= line is caller's preferred sending codec
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()?;
Some((chosen.to_params(), dtmf, chosen.codec))
});
from_offer.unwrap_or_else(|| {
// Fallback: use answer SDP
self.answer
.as_ref()
.map(|s| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(s);
let chosen = codecs
.iter()
.find(|c| c.codec == codec_b)
.cloned()
.or_else(|| {
codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()
})
.unwrap_or(CodecInfo {
payload_type: 0,
codec: CodecType::PCMU,
clock_rate: 8000,
channels: 1,
});
(chosen.to_params(), dtmf, chosen.codec)
})
.unwrap_or((
rustrtc::RtpCodecParameters::default(),
None,
CodecType::PCMU,
))
})
let caller_is_webrtc = self
.caller_offer
.as_ref()
.map(|offer| Self::is_webrtc_sdp(offer))
.unwrap_or(false);
let from_offer = self.caller_offer.as_ref().and_then(|offer| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(offer);
let chosen = codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()?;
Some((chosen.to_params(), dtmf, chosen.codec))
});
let from_answer = self.answer.as_ref().map(|s| {
let (codecs, dtmf) = MediaNegotiator::extract_codec_params(s);
let chosen = codecs
.iter()
.find(|c| c.codec == codec_b)
.cloned()
.or_else(|| {
codecs
.iter()
.find(|c| c.codec != CodecType::TelephoneEvent)
.cloned()
})
.unwrap_or(CodecInfo {
payload_type: 0,
codec: CodecType::PCMU,
clock_rate: 8000,
channels: 1,
});
(chosen.to_params(), dtmf, chosen.codec)
});
let default_codec = (
rustrtc::RtpCodecParameters::default(),
None,
CodecType::PCMU,
);
let (params_a, dtmf_pt_a, codec_a) = if caller_is_webrtc {
from_answer.or(from_offer).unwrap_or(default_codec)
} else {
from_offer.or(from_answer).unwrap_or(default_codec)
};

let ssrc_a = self
Expand Down