freeswitch-webrtc-bench

WebRTC benchmark for FreeSWITCH.

About

Freeswitch-webrtc-bench is a loading test tool like sipp and winsip, but mainly focus on SIP signalling over websocket and media via WebRTC. It is built on go-sip-ua/gosip/pion/webrtc and is pure Go.

There is a web client rztrtcsdk created by me that you can use to test WebRTC with freeswitch-webrtc-bench.

Usage

• Compile

  go build .

• Create a call profile

  There are some example call profiles in the examples directory that you can modify with reference to the Call Profile description.

• Run the test

  Run the UAS first,

  ./freeswitch-webrtc-bench -p examples/uas.json

  Then in another console, run the UAC,

  ./freeswitch-webrtc-bench -p examples/uac.json

  After the test is done, there will be some log files generated.

Call Profile

Here is the call profile structure defined in Go. You can find some example call profiles in the examples directory.

// CallProfile is a json config file for UAC and UAS
type CallProfile struct {
	// "file" or "console"; the log file name looks like uac_benchlog_202103261437.log
	LogType string `json:"logtype"`
	// log directory
	LogDir string `json:"logdir"`
	// put call details in log
	CallDetailsInLog bool `json:"calldetailsinlog"`
	// separate call details file; the call details file name looks like uac_calldetails_202103261437.json
	CallDetailsFile bool `json:"calldetailsfile"`
	// put sdp in call details
	SdpInCallDetails bool `json:"sdpincalldetails"`

	// "0.0.0.0" or a specified IP; similar to '-i' of sipp
	LocalIP string `json:"localip"`
	// "0" or a specified port range which must be the same length as the accounts; similar to '-p' of sipp
	//
	// doesn't support all calls share one port
	LocalPorts         string   `json:"localports"`
	InternalLocalPorts []string // `json:"internallocalports,omitempty"`
	// "udp", "tcp", "tls", "ws" or "wss"; similar to '-t' of sipp
	Transport string `json:"transport"`

	ServerIP   string `json:"serverip"`
	ServerPort string `json:"serverport"`
	// comma separated ICE URLs
	ICEURLs       string `json:"iceurls"`
	ICEUsername   string `json:"iceusername"`
	ICECredential string `json:"icecredential"`

	// "uac" or "uas"; similar to '-sn' of sipp
	Role string `json:"role"`
	// an account range
	Accounts         string   `json:"accounts"`
	InternalAccounts []string // `json:"internalaccounts,omitempty"`
	// a password range, it should have the same length as the accounts
	Passwords         string   `json:"passwords"`
	InternalPasswords []string // `json:"internalpasswords,omitempty"`
	// for uas, it is ""
	//
	// for uac, it is a called party range, it should have the same length as the accounts
	CalledParties         string   `json:"calledparties"`
	InternalCalledParties []string // `json:"internalcalledparties,omitempty"`

	// maximum number of concurrent calls; similar to '-l' of sipp
	//
	// it can't be greater than the length of the accounts; if <= 0, it will be set to the length of the accounts
	MaxConcurrentCalls int `json:"maxconcurrentcalls"`
	// total number of processed calls; similar to '-m' of sipp
	//
	// after total calls, program exits; if <= 0, no totalcalls limitation
	TotalCalls int `json:"totalcalls"`
	// how long to send bye after the call is established, unit is millisecond
	//
	// if <= 0, no callduration limitation; similar to '-d' of sipp
	CallDuration int `json:"callduration"`
	// calls initiated every batch; similar to '-r' of sipp
	CallsPerBatch int `json:"callsperbatch"`
	// interval between every batch, unit is millisecond; similar to '-rp' of sipp
	BatchInterval int `json:"batchinterval"`

	// "audio" or "video" for uac
	//
	// the mediatype of uas depends on received SDP
	MediaType string `json:"mediatype"`
	// file path of the sending audio file
	//
	// it can be an absolute path or a relative path to the call profile
	AudioFile string `json:"audiofile"`
	// file path of the sending video file
	VideoFile string `json:"videofile"`
	// fps of the sending video file, default is 15
	VideoFileFps int `json:"videofilefps"`

	// "true" or "false"
	//
	// only the first call will dump the media
	DumpMedia bool `json:"dumpmedia"`
	// file path of the dumping audio file without suffix
	//
	// it can be an absolute path or a relative path to the call profile
	SavedAudioFile string `json:"savedaudiofile"`
	// file path of the dumping video file without suffix
	SavedVideoFile string `json:"savedvideofile"`
}

Notes

There is an issue Change findMatchingCipherSuite when pion/dtls and FreeSWITCH interoperate. I've modified the code in the vendor directory, but if you run go mod vendor command, you might need to modify it again manually.

diff --git a/vendor/github.com/pion/dtls/v2/util.go b/vendor/github.com/pion/dtls/v2/util.go
index 745182d..cd7249d 100644
--- a/vendor/github.com/pion/dtls/v2/util.go
+++ b/vendor/github.com/pion/dtls/v2/util.go
@@ -11,9 +11,9 @@ func findMatchingSRTPProfile(a, b []SRTPProtectionProfile) (SRTPProtectionProfil
        return 0, false
 }

-func findMatchingCipherSuite(a, b []CipherSuite) (CipherSuite, bool) { //nolint
-       for _, aSuite := range a {
-               for _, bSuite := range b {
+func findMatchingCipherSuite(remote, local []CipherSuite) (CipherSuite, bool) { //nolint
+       for _, aSuite := range local {
+               for _, bSuite := range remote {
                        if aSuite.ID() == bSuite.ID() {
                                return aSuite, true
                        }

Limits

• Only supports one media

  • Since FreeSWITCH doesn't support bundled media, and pion/webrtc doesn't support unbundled media, so there is only one media in a call, either audio or video, you can specify it by setting the mediatype parameter in the call profile.

• Audio call is not fully supported

  • Audio call can be established now; but if the remote party rejects us with 603 Decline, FreeSWITCH won't hang up us.
  • The remote party cannot hear us, because FreeSWITCH doesn't pass our media to the remote party; but we can hear the remote party.

• Call without TURN enabled is not supported

  • UAS ICE state failed if TURN is not enabled. That might be pion's problem (TODO).

Acknowledgement

• cloudwebrtc/go-sip-ua SIP UA
• ghettovoice/gosip SIP Stack
• pion/webrtc Pion WebRTC
• ossrs/srs-bench SRS WebRTC Benchmark