move openbsc/* to repos root

This is the first step in creating this repository from the legacy openbsc.git.

Like all other Osmocom repositories, keep the autoconf and automake files in
the repository root. openbsc.git has been the sole exception, which ends now.

Change-Id: I9c6f2a448d9cb1cc088cf1cf6918b69d7e69b4e7

1 files changed, 87 insertions, 0 deletions

diff --git a/contrib/rtp/rtp_replay_sip.st b/contrib/rtp/rtp_replay_sip.st
new file mode 100644
index 000000000..5f844df1d
--- /dev/null
+++ b/contrib/rtp/rtp_replay_sip.st
@@ -0,0 +1,87 @@
+"""
+Create a SIP connection and then stream...
+"""
+
+PackageLoader
+    fileInPackage: #OsmoSIP.
+
+"Load for the replay code"
+FileStream fileIn: 'rtp_replay_shared.st'.
+
+
+Osmo.SIPCall subclass: StreamCall [
+    | sem stream |
+
+    createCall: aSDP [
+        | sdp |
+        stream := RTPReplay on: 'rtp_ssrc6976010.240.240.1_to_10.240.240.50.state'.
+        sdp := aSDP % {stream localPort}.
+        ^ super createCall: sdp.
+    ]
+
+    sem: aSemaphore [
+          sem := aSemaphore
+    ]
+
+    sessionNew [
+        | host port |
+        Transcript nextPutAll: 'The call has started'; nl.
+        Transcript nextPutAll: sdp_result; nl.
+
+        host := SDPUtils findHost: sdp_result.
+        port := SDPUtils findPort: sdp_result.
+
+        [
+            stream streamAudio: host port: port.
+            Transcript nextPutAll: 'Streaming has finished.'; nl.
+        ] fork.
+    ]
+
+    sessionFailed [
+        sem signal
+    ]
+
+    sessionEnd [
+        sem signal
+    ]
+]
+
+Eval [
+    | transport agent call sem sdp_fr sdp_amr |
+
+
+    sdp_fr := (WriteStream on: String new)
+        nextPutAll: 'v=0'; cr; nl;
+        nextPutAll: 'o=twinkle 1739517580 1043400482 IN IP4 127.0.0.1'; cr; nl;
+        nextPutAll: 's=-'; cr; nl;
+        nextPutAll: 'c=IN IP4 127.0.0.1'; cr; nl;
+        nextPutAll: 't=0 0'; cr; nl;
+        nextPutAll: 'm=audio %1 RTP/AVP 0 101'; cr; nl;
+        nextPutAll: 'a=rtpmap:0 PCMU/8000'; cr; nl;
+        nextPutAll: 'a=rtpmap:101 telephone-event/8000'; cr; nl;
+        nextPutAll: 'a=fmtp:101 0-15'; cr; nl;
+        nextPutAll: 'a=ptime:20'; cr; nl;
+        contents.
+
+    sem := Semaphore new.
+    transport := Osmo.SIPUdpTransport
+          startOn: '0.0.0.0' port: 5066.
+    agent := Osmo.SIPUserAgent createOn: transport.
+    transport start.
+
+    call := (StreamCall
+              fromUser: 'sip:1000@sip.zecke.osmocom.org'
+              host: '127.0.0.1'
+              port: 5060
+              to: 'sip:123456@127.0.0.1'
+              on: agent)
+              sem: sem; yourself.
+
+    call createCall: sdp_fr.
+
+
+    "Wait for the stream to have ended"
+    sem wait.
+
+    (Delay forSeconds: 4) wait.
+]