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Adding ability to specify per-trunk media timeouts #1534
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| @@ -0,0 +1,5 @@ | ||
| --- | ||
| "github.com/livekit/protocol": minor | ||
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| Adding ability to specify media timeout per SIP trunk |
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@@ -355,7 +355,10 @@ message SIPInboundTrunkInfo { | |
| google.protobuf.Timestamp created_at = 17; | ||
| google.protobuf.Timestamp updated_at = 18; | ||
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| // NEXT ID: 19 | ||
| // Max time a call can last without incoming RTP data received. If unset, defaults are used. | ||
| google.protobuf.Duration media_timeout = 19; | ||
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Contributor
There was a problem hiding this comment. Choose a reason for hiding this commentThe reason will be displayed to describe this comment to others. Learn more. Lets move it to the dispatch rule level, similar to #1530. If you move this field to the In general we should avoid adding new settings to trunks, as they might be gone after we implement the webhooks. |
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| // NEXT ID: 20 | ||
| } | ||
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| message SIPInboundTrunkUpdate { | ||
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@@ -370,6 +373,7 @@ message SIPInboundTrunkUpdate { | |
| (logger.redact_format) = "<redacted ({{ .Size }} bytes)>" | ||
| ]; | ||
| optional SIPMediaEncryption media_encryption = 8; | ||
| optional google.protobuf.Duration media_timeout = 9; | ||
| } | ||
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| message CreateSIPOutboundTrunkRequest { | ||
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@@ -443,7 +447,10 @@ message SIPOutboundTrunkInfo { | |
| google.protobuf.Timestamp created_at = 16; | ||
| google.protobuf.Timestamp updated_at = 17; | ||
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| // NEXT ID: 18 | ||
| // Max time a call can last without incoming RTP data received. If unset, defaults are used. | ||
| google.protobuf.Duration media_timeout = 18; | ||
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Contributor
There was a problem hiding this comment. Choose a reason for hiding this commentThe reason will be displayed to describe this comment to others. Learn more. Same here, we could just leave it on the |
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| // NEXT ID: 19 | ||
| } | ||
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| message SIPOutboundTrunkUpdate { | ||
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@@ -460,8 +467,9 @@ message SIPOutboundTrunkUpdate { | |
| ]; | ||
| optional SIPMediaEncryption media_encryption = 8; | ||
| optional string from_host = 10; | ||
| optional google.protobuf.Duration media_timeout = 11; | ||
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| // NEXT ID: 11 | ||
| // NEXT ID: 12 | ||
| } | ||
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| message GetSIPInboundTrunkRequest { | ||
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@@ -716,7 +724,10 @@ message SIPOutboundConfig { | |
| // Optional custom hostname for the 'From' SIP header. When set, outbound calls use this host instead of the default project SIP domain. | ||
| string from_host = 8; | ||
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| // NEXT ID: 9 | ||
| // Max time a call can last without incoming RTP data received. If unset, defaults are used. | ||
| google.protobuf.Duration media_timeout = 9; | ||
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| // NEXT ID: 10 | ||
| } | ||
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| // A SIP Participant is a singular SIP session connected to a LiveKit room via | ||
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@@ -240,7 +240,10 @@ message EvaluateSIPDispatchRulesResponse { | |
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| map<string, string> feature_flags = 23; | ||
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| // NEXT ID: 24 | ||
| // Per-call RTP media timeout; if unset, SIP service defaults apply. | ||
| google.protobuf.Duration media_timeout = 24; | ||
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Contributor
There was a problem hiding this comment. Choose a reason for hiding this commentThe reason will be displayed to describe this comment to others. Learn more. Do we want to allow overriding the initial media timeout as well? |
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| // NEXT ID: 25 | ||
| } | ||
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| message UpdateSIPCallStateRequest { | ||
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After #1530 you could use
new(MaxSIPMediaTimeout)as it updates Go to 1.26.