feat: join and WebRTC connection latency DAGs

[tbarbugli]

Why

VOICE_LATENCY_DAG=1 (#674) shows where a turn's time goes, but not where the time before the first turn goes. Joining a call is several hops: the agent creates the call on the coordinator, opens the router session and event socket, and the router then joins the call and negotiates WebRTC (coordinator and SFU sockets and joins, offer/answer, ICE, DTLS, first RTP). Slow starts were hard to attribute without seeing each of those steps, so this prints them as DAGs alongside the existing turn DAG.

Depends on GetStream/getstream-go-webrtc#2, which adds Call.OnConnectionTiming. Until that merges, acceleration/go.mod pins getstream-go-webrtc to its branch feature/connection-timing at 0ab2ab2; the pin should move to the merged commit before this lands.

Changes

• The Stream edge reports, per peer connection (publisher and subscriber), every step from the start of its join, including SFU offer received, SetPublisher / SendAnswer, ICE start and checks, DTLS and first RTP. A path is reported once it has its first packet, or after a short wait without one.
• The agent logs each path's timing and emits it as an event; the session socket forwards it as a connection frame, and a watcher that attaches after the path connected is sent the latest timing.
• The Python agent records its own join steps (create call, components, chat conversation, router session, event socket), passes them to the pipeline as RemoteCall.join_steps, and stream.Accelerated prints a join DAG and a WebRTC connection DAG per connection frame when log_latency is on.
• The streamedge join conflict with 1480919 is resolved by registering the timing callback on the local call before Join, keeping that fix's "only set e.call after a successful join".

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