Refactor call classes

README.md

@@ -47,7 +47,7 @@ SIP_PASSWORD=******** uvx 'voip[cli]' sip sips:alice@sip.example.com agent
 uv add voip[audio,ai,pygments]
 ```
 
-Subclass `WhisperCall` and override `transcription_received` to handle results.
+Subclass `TranscribeCall` and override `transcription_received` to handle results.
 Pass it as `call_class` when answering an incoming call:
 
 ```python

docs/calls.md

@@ -6,15 +6,11 @@
 
 ::: voip.audio.AudioCall
 
-## Voice Activity Detection
-
 ::: voip.audio.VoiceActivityCall
 
-## Echo Call
-
 ::: voip.audio.EchoCall
 
-## AI / Agentic Calls
+## AI Calls
 
 ::: voip.ai.TranscribeCall
 

docs/cookbook.md

@@ -76,14 +76,13 @@ from pocket_tts import TTSModel
 from voip.ai import AgentCall
 from voip.sip.protocol import SIP
 
-
 shared_tts = TTSModel.load_model()
 
 
 class MyCall(AgentCall):
     tts_model = shared_tts
     system_prompt = "You are a friendly hotel receptionist. Keep answers brief."
-    ollama_model = "llama3"
+    llm_model = "llama3"
     voice = "azelma"
 
 

tests/sip/test_protocol.py

@@ -495,7 +495,7 @@ async def _run_answer(self, protocol, invite, fake_rtp_transport):
         protocol._rtp_transport = fake_rtp_transport
         # Resolve the SIP protocol's own local address (for Contact header).
         protocol.local_address = ("127.0.0.1", 5061)
-        await protocol._answer(invite, _CodecAwareCall)
+        await protocol.answer(invite, call_class=_CodecAwareCall)
 
     @pytest.mark.asyncio
     async def test_answer__selects_pcma_from_offer(self, fake_rtp_transport):
@@ -662,7 +662,7 @@ async def test_answer__no_address_logs_error(self, caplog):
         invite = self._make_invite("no-addr-answer-1")
 
         with caplog.at_level("ERROR"):
-            await protocol._answer(invite, RTPCall)
+            await protocol.answer(invite, call_class=RTPCall)
         assert "No pending INVITE found" in caplog.text
         assert not protocol._sent_responses
 
@@ -1116,7 +1116,7 @@ async def test_answer__sends_200_ok(self):
         protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
-        await protocol._answer(request, RTPCall)
+        await protocol.answer(request, call_class=RTPCall)
         assert len(protocol._sent) == 1
         response, _ = protocol._sent[0]
         assert response.status_code == 200
@@ -1137,7 +1137,7 @@ async def test_answer__sdp_contains_opus_audio_line(self):
         protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
-        await protocol._answer(request, RTPCall)
+        await protocol.answer(request, call_class=RTPCall)
         response, _ = protocol._sent[0]
         assert b"m=audio" in bytes(response.body)
         assert b"RTP/SAVP 0" in bytes(response.body)
@@ -1183,9 +1183,9 @@ async def test_answer__rtp_avp_offer_returns_rtp_avp(self):
 
         request = Request.parse(invite_bytes)
         protocol._pending_invites.add(request.headers["Call-ID"])
-        AudioCall = pytest.importorskip("voip.audio.AudioCall")
+        AudioCall = pytest.importorskip("voip.audio").AudioCall
 
-        await protocol._answer(request, AudioCall)
+        await protocol.answer(request, call_class=AudioCall)
         response, _ = protocol._sent[0]
         body = bytes(response.body)
         assert b"RTP/AVP" in body
@@ -1221,9 +1221,9 @@ async def test_answer__rtp_savp_offer_returns_rtp_savp(self):
 
         request = Request.parse(invite_bytes)
         protocol._pending_invites.add(request.headers["Call-ID"])
-        AudioCall = pytest.importorskip("voip.audio.AudioCall")
+        AudioCall = pytest.importorskip("voip.audio").AudioCall
 
-        await protocol._answer(request, AudioCall)
+        await protocol.answer(request, call_class=AudioCall)
         response, _ = protocol._sent[0]
         body = bytes(response.body)
         assert b"RTP/SAVP" in body
@@ -1247,7 +1247,7 @@ async def test_answer__copies_dialog_headers(self):
         protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
-        await protocol._answer(request, RTPCall)
+        await protocol.answer(request, call_class=RTPCall)
         response, _ = protocol._sent[0]
         assert response.headers["Via"] == "SIP/2.0/UDP pc33.atlanta.com"
         assert response.headers["To"] == "sip:alice@atlanta.com"
@@ -1277,7 +1277,7 @@ def __post_init__(self) -> None:
         protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
-        await protocol._answer(request, MyCall)
+        await protocol.answer(request, call_class=MyCall)
         assert created == ["sip:bob@biloxi.com"]
 
     async def test_answer__rtp_receives_audio(self):
@@ -1304,7 +1304,7 @@ def packet_received(self, packet: RTPPacket, addr) -> None:
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
         try:
-            await protocol._answer(request, PacketCapture)
+            await protocol.answer(request, call_class=PacketCapture)
             response, _ = protocol._sent[0]
             sdp_line = next(
                 line
@@ -1352,7 +1352,7 @@ def packet_received(self, packet: RTPPacket, addr) -> None:
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
         try:
-            await protocol._answer(request, PacketCapture)
+            await protocol.answer(request, call_class=PacketCapture)
             response, _ = protocol._sent[0]
             sdp_line = next(
                 line
@@ -1391,7 +1391,7 @@ async def test_answer__content_length_serialized(self):
         protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
-        await protocol._answer(request, RTPCall)
+        await protocol.answer(request, call_class=RTPCall)
         response, _ = protocol._sent[0]
         serialized = bytes(response)
         parsed = Message.parse(serialized)
@@ -1429,7 +1429,7 @@ async def test_answer__reuses_shared_rtp_socket_for_second_call(self):
             body=sdp_body1,
         )
         protocol._pending_invites.add("call-1@test")
-        await protocol._answer(invite1, _MinimalCall)
+        await protocol.answer(invite1, call_class=_MinimalCall)
         rtp_proto_1 = protocol._rtp_protocol
         rtp_transport_1 = protocol._rtp_transport
 
@@ -1449,7 +1449,7 @@ async def test_answer__reuses_shared_rtp_socket_for_second_call(self):
             body=sdp_body2,
         )
         protocol._pending_invites.add("call-2@test")
-        await protocol._answer(invite2, _MinimalCall)
+        await protocol.answer(invite2, call_class=_MinimalCall)
 
         assert protocol._rtp_protocol is rtp_proto_1
         assert protocol._rtp_transport is rtp_transport_1
@@ -1473,7 +1473,7 @@ async def test_answer__bye_unregisters_call_from_rtp_mux(self):
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
         try:
-            await protocol._answer(request, RTPCall)
+            await protocol.answer(request, call_class=RTPCall)
             assert None in mux.calls
 
             bye = Request(
@@ -1510,7 +1510,7 @@ async def test_answer__logs_info(self, caplog):
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
         with caplog.at_level(logging.INFO, logger="voip.sip"):
-            await protocol._answer(request, RTPCall)
+            await protocol.answer(request, call_class=RTPCall)
         assert any("call_answered" in r.message for r in caplog.records)
 
     def test_reject__sends_busy_here_by_default(self):
@@ -1592,26 +1592,30 @@ async def test_request_received__unsupported_method__raises(self):
 
     async def test_answer__via_call_received__schedules_answer(self):
         """answer() is async; wrapping it in create_task from call_received works."""
-        answered = []
 
-        class MySIP(SIP):
+        class MySIP(self._CapturingSIP):
             def call_received(self, request):
                 asyncio.create_task(
                     self.answer(request=request, call_class=_MinimalCall)
                 )
 
-            async def _answer(self, request, call_class):
-                answered.append((request, call_class))
-
-        protocol = MySIP(outbound_proxy=("127.0.0.1", 5060), aor="sip:test@example.com")
-        protocol.connection_made(MagicMock())
+        loop = asyncio.get_running_loop()
+        protocol = MySIP()
+        protocol.transport = make_mock_transport()
+        protocol.local_address = ("127.0.0.1", 5061)
+        mux = RealtimeTransportProtocol()
+        mux.public_address = loop.create_future()
+        mux.public_address.set_result(("127.0.0.1", 12000))
+        mock_rtp_transport = MagicMock()
+        mock_rtp_transport.get_extra_info.return_value = ("127.0.0.1", 12000)
+        protocol._rtp_protocol = mux
+        protocol._rtp_transport = mock_rtp_transport
         request = make_invite()
         protocol._pending_invites.add(request.headers["Call-ID"])
         protocol.call_received(request)
 
-        await asyncio.sleep(0.01)
-        assert len(answered) == 1
-        assert answered[0][1] is _MinimalCall
+        await asyncio.sleep(0.05)
+        assert len(protocol._sent) == 1
 
 
 # ---------------------------------------------------------------------------