contrib: Create a script that opens a SIP session before

Use the Smalltalk SIP implementation to create a call
and once the call has been established start the replay
using the commoncode. No patching of RTP occurs yet.

2 files changed, 130 insertions, 4 deletions

diff --git a/openbsc/contrib/rtp/rtp_replay_shared.st b/openbsc/contrib/rtp/rtp_replay_shared.st
index cfb66bb4d..dd32aed7a 100644
--- a/openbsc/contrib/rtp/rtp_replay_shared.st
+++ b/openbsc/contrib/rtp/rtp_replay_shared.st
@@ -4,19 +4,58 @@ Simple UDP replay from the state files
 
 PackageLoader fileInPackage: #Sockets.
 
+Object subclass: SDPUtils [
+    "Look into using PetitParser."
+    SDPUtils class >> findPort: aSDP [
+        aSDP linesDo: [:line |
+            (line startsWith: 'm=audio ') ifTrue: [
+                | stream |
+                stream := line readStream
+                            skip: 'm=audio ' size;
+                            yourself.
+                ^ Number readFrom: stream.
+            ]
+        ].
+
+        ^ self error: 'Not found'.
+    ]
+
+    SDPUtils class >> findHost: aSDP [
+        aSDP linesDo: [:line |
+            (line startsWith: 'c=IN IP4 ') ifTrue: [
+                | stream |
+                ^ stream := line readStream
+                            skip: 'c=IN IP4 ' size;
+                            upToEnd.
+            ]
+        ].
+
+        ^ self error: 'Not found'.
+    ]
+]
+
 Object subclass: RTPReplay [
-    | filename |
+    | filename socket |
     RTPReplay class >> on: aFile [
         ^ self new
+            initialize;
             file: aFile; yourself
     ]
 
+    initialize [
+        socket := Sockets.DatagramSocket new.
+    ]
+
     file: aFile [ 
         filename := aFile
     ]
 
+    localPort [
+        ^ socket port
+    ]
+
     streamAudio: aHost port: aPort [
-        | file last_time last_image udp_send socket dest |
+        | file last_time last_image udp_send dest |
 
         last_time := nil.
         last_image := nil.
@@ -24,7 +63,6 @@ Object subclass: RTPReplay [
 
         "Send the payload"
         dest := Sockets.SocketAddress byName: aHost.
-        socket := Sockets.DatagramSocket new.
         udp_send := [:payload | | datagram |
             datagram := Sockets.Datagram data: payload contents address: dest port: aPort.
             socket nextPut: datagram
@@ -57,7 +95,8 @@ Object subclass: RTPReplay [
 
                     "How long to wait?"
                     wait_image := last_image + ((time - last_time) * 1000).
-                    [ wait_image > Time millisecondClockValue ] whileTrue: [].
+                    [ wait_image > Time millisecondClockValue ]
+                        whileTrue: [Processor yield].
 
                     udp_send value: data.
                     last_time := time.
diff --git a/openbsc/contrib/rtp/rtp_replay_sip.st b/openbsc/contrib/rtp/rtp_replay_sip.st
new file mode 100644
index 000000000..5f844df1d
--- /dev/null
+++ b/openbsc/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.
+]