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Add support for turn logging id in ios sdk.

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This patch adds support for setting the TURN_LOGGING_ID
in RTCConfig using the ios SDK.

TURN_LOGGING_ID was added to webrtc in
https://webrtc-review.googlesource.com/c/src/+/149829

The intended usage of this attribute is to correlate client and
backend logs.

This change was tested out with duo via wireshark.

Bug: webrtc:10897
Change-Id: Iedbefdc6392c4df203aca08cf750028b450a11ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191340
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Brad Pugh <bradpugh@google.com>
Cr-Commit-Position: refs/heads/master@{#32626}
This commit is contained in:
Brad Pugh 2020-11-16 16:27:20 -08:00 committed by Commit Bot
parent 97050115f0
commit f24143d3b0
2 changed files with 9 additions and 0 deletions
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6
sdk/objc/api/peerconnection/RTCConfiguration.h
View File

@ -204,6 +204,12 @@ RTC_OBJC_EXPORT
*/
@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
/**
* An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
* which can be used to correlate client logs with backend logs.
*/
@property(nonatomic, nullable, copy) NSString *turnLoggingId;
/**
* Time interval between audio RTCP reports.
*/

3
sdk/objc/api/peerconnection/RTCConfiguration.mm
View File

@ -53,6 +53,7 @@
@synthesize activeResetSrtpParams = _activeResetSrtpParams;
@synthesize allowCodecSwitching = _allowCodecSwitching;
@synthesize cryptoOptions = _cryptoOptions;
@synthesize turnLoggingId = _turnLoggingId;
@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
@ -129,6 +130,7 @@
sframeRequireFrameEncryption:config.crypto_options->sframe
.require_frame_encryption];
}
_turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
_rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
_rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
_allowCodecSwitching = config.allow_codec_switching.value_or(false);
@ -258,6 +260,7 @@
_cryptoOptions.sframeRequireFrameEncryption ? true : false;
nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
}
nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
nativeConfig->allow_codec_switching = _allowCodecSwitching;