diff --git a/plugins/sip/src/rtp-utils.ts b/plugins/sip/src/rtp-utils.ts
index b7bb0651b..1c48dcc17 100644
--- a/plugins/sip/src/rtp-utils.ts
+++ b/plugins/sip/src/rtp-utils.ts
@@ -11,6 +11,7 @@ export interface RtpStreamOptions {
 
 export interface RtpOptions {
   audio: RtpStreamOptions
+  video: RtpStreamOptions
 }
 
 export interface RtpStreamDescription extends RtpStreamOptions {
@@ -22,6 +23,7 @@ export interface RtpStreamDescription extends RtpStreamOptions {
 export interface RtpDescription {
   address: string
   audio: RtpStreamDescription
+  video: RtpStreamDescription
   sdp: string
 }
 
@@ -52,10 +54,12 @@ export function sendStunBindingRequest({
   rtcpSplitter: dgram.Socket
   rtpDescription: RtpDescription
   localUfrag?: string
-  type: 'video' | 'audio'
+  type: 'audio' | 'video'
 }) {
+  const  remoteDescription = rtpDescription[type];
+  if( remoteDescription.port == 0 )
+    return;
   const message = stun.createMessage(1),
-    remoteDescription = rtpDescription[type],
     { address } = rtpDescription,
     { iceUFrag, icePwd, port, rtcpPort } = remoteDescription
 
diff --git a/plugins/sip/src/sip-call.ts b/plugins/sip/src/sip-call.ts
index 4225a797f..a5e1e2346 100644
--- a/plugins/sip/src/sip-call.ts
+++ b/plugins/sip/src/sip-call.ts
@@ -10,6 +10,8 @@ export interface SipOptions {
   from: string
   localIp: string
   localPort: number
+  udp: boolean
+  tcp: boolean
 }
 
 interface UriOptions {
@@ -61,17 +63,24 @@ function getRandomId() {
 function getRtpDescription(
   console: any,
   sections: string[],
-  mediaType: 'audio'
+  mediaType: 'audio' | 'video'
 ): RtpStreamDescription {
   try {
-    const section = sections.find((s) => s.startsWith('m=' + mediaType)),
-      { port } = sdp.parseMLine(section),
-      lines: string[] = sdp.splitLines(section),
-      rtcpLine = lines.find((l: string) => l.startsWith('a=rtcp:')),
-      rtcpMuxLine = lines.find((l: string) => l.startsWith('a=rtcp-mux')),
-      ssrcLine = lines.find((l: string) => l.startsWith('a=ssrc')),
-      iceUFragLine = lines.find((l: string) => l.startsWith('a=ice-ufrag')),
-      icePwdLine = lines.find((l: string) => l.startsWith('a=ice-pwd'))
+    const section = sections.find((s) => s.startsWith('m=' + mediaType));
+    if( section === undefined ) {
+      return {
+        port: 0,
+        rtcpPort: 0
+      };
+    }
+
+    const { port } = sdp.parseMLine(section),
+    lines: string[] = sdp.splitLines(section),
+    rtcpLine = lines.find((l: string) => l.startsWith('a=rtcp:')),
+    rtcpMuxLine = lines.find((l: string) => l.startsWith('a=rtcp-mux')),
+    ssrcLine = lines.find((l: string) => l.startsWith('a=ssrc')),
+    iceUFragLine = lines.find((l: string) => l.startsWith('a=ice-ufrag')),
+    icePwdLine = lines.find((l: string) => l.startsWith('a=ice-pwd'))
 
     let rtcpPort: number;
     if (rtcpMuxLine) {
@@ -105,7 +114,8 @@ function parseRtpDescription(console: any, inviteResponse: {
   return {
     sdp: inviteResponse.content,
     address: cLine.match(/c=IN IP4 (\S*)/)![1],
-    audio: getRtpDescription(console, sections, 'audio')
+    audio: getRtpDescription(console, sections, 'audio'),
+    video: getRtpDescription(console, sections, 'video')
   }
 }
 
@@ -131,7 +141,7 @@ export class SipCall {
   ) {
     this.console = console;
 
-    const { audio } = rtpOptions,
+    const { audio, video } = rtpOptions,
       { from } = this.sipOptions,
       host = this.sipOptions.localIp,
       port = this.sipOptions.localPort,
@@ -143,13 +153,13 @@ export class SipCall {
         host,
         hostname: host,
         port: port,
-        udp: true,
-        tcp: false,
+        udp: this.sipOptions.udp,
+        tcp: this.sipOptions.tcp,
         tls: false,
         // tls_port: tlsPort,
         // tls: {
         //   rejectUnauthorized: false,
-        // },        
+        // },
         ws: false
       },
         (request: SipRequest) => {
@@ -179,6 +189,25 @@ export class SipCall {
     ]
       .filter((l) => l)
       .join('\r\n')) + '\r\n';
+
+      /* Example SDP for audio and video
+    this.sdp = ([
+      'v=0',
+      `o=${from.split(':')[1].split('@')[0]} 3747 461 IN IP4 ${host}`,
+      's=ScryptedSipPlugin',
+      `c=IN IP4 ${host}`,
+      't=0 0',
+      `m=audio ${audio.port} RTP/AVP 97`,
+      `a=rtpmap:97 speex/8000`,
+      `m=video ${video.port} RTP/AVP 97`,
+      `a=rtpmap:97 H264/90000`,
+      `a=fmtp:97 profile-level-id=42801F`,
+      `a=ssrc:${ssrc}`,
+      'a=recvonly'
+    ]
+      .filter((l) => l)
+      .join('\r\n')) + '\r\n';
+      */
   }
 
   request({
@@ -299,7 +328,9 @@ export class SipCall {
   }
 
   destroy() {
+    this.console.debug("detroying sip-call")
     this.destroyed = true
     this.sipStack.destroy()
+    this.console.debug("detroying sip-call: done")
   }
 }
diff --git a/plugins/sip/src/sip-session.ts b/plugins/sip/src/sip-session.ts
index d9963e9f2..c65b6e9ac 100644
--- a/plugins/sip/src/sip-session.ts
+++ b/plugins/sip/src/sip-session.ts
@@ -1,5 +1,5 @@
 import { reservePorts } from '@homebridge/camera-utils';
-import { createBindUdp } from '@scrypted/common/src/listen-cluster';
+import { createBindUdp, createBindZero } from '@scrypted/common/src/listen-cluster';
 import dgram from 'dgram';
 import { ReplaySubject, timer } from 'rxjs';
 import { createStunResponder, RtpDescription, RtpOptions, sendStunBindingRequest } from './rtp-utils';
@@ -19,6 +19,8 @@ export class SipSession extends Subscribed {
     public readonly rtpOptions: RtpOptions,
     public readonly audioSplitter: dgram.Socket,
     public audioRtcpSplitter: dgram.Socket,
+    public readonly videoSplitter: dgram.Socket,
+    public videoRtcpSplitter: dgram.Socket,
     public readonly cameraName: string
   ) {
     super()
@@ -27,13 +29,18 @@ export class SipSession extends Subscribed {
   }
 
   static async createSipSession(console: any, cameraName: string, sipOptions: SipOptions) {
-    const audioPort = 0,
-      audioSplitter = await createBindUdp(audioPort),
+      const audioSplitter = await createBindZero(),
       audioRtcpSplitter = await createBindUdp(audioSplitter.port + 1),
+      videoSplitter = await createBindZero(),
+      videoRtcpSplitter = await createBindUdp(videoSplitter.port + 1),
       rtpOptions = {
         audio: {
           port: audioSplitter.port,
           rtcpPort: audioRtcpSplitter.port
+        },
+        video: {
+          port: videoSplitter.port,
+          rtcpPort: videoRtcpSplitter.port
         }
       }
 
@@ -43,6 +50,8 @@ export class SipSession extends Subscribed {
       rtpOptions,
       audioSplitter.server,
       audioRtcpSplitter.server,
+      videoSplitter.server,
+      videoRtcpSplitter.server,
       cameraName
     )
   }
@@ -87,18 +96,35 @@ export class SipSession extends Subscribed {
             localUfrag: this.sipCall.audioUfrag,
             type: 'audio',
           })
+          sendStunBindingRequest({
+            rtpSplitter: this.videoSplitter,
+            rtcpSplitter: this.videoRtcpSplitter,
+            rtpDescription,
+            localUfrag: this.sipCall.videoUfrag,
+            type: 'video',
+          })
         }
 
       // if rtcp-mux is supported, rtp splitter will be used for both rtp and rtcp
-      if (rtpDescription.audio.port === rtpDescription.audio.rtcpPort) {
+      if ( rtpDescription.audio.port > 0 && rtpDescription.audio.port === rtpDescription.audio.rtcpPort) {
         this.audioRtcpSplitter.close()
         this.audioRtcpSplitter = this.audioSplitter
       }
 
-      if (rtpDescription.audio.iceUFrag) {
+      if ( rtpDescription.video.port > 0 && rtpDescription.video.port === rtpDescription.video.rtcpPort) {
+        this.videoRtcpSplitter.close()
+        this.videoRtcpSplitter = this.videoSplitter
+      }
+
+      if ( (rtpDescription.audio.port > 0 && rtpDescription.audio.iceUFrag)|| (rtpDescription.video.port > 0 && rtpDescription.video.iceUFrag ) ) {
         // ICE is supported
         this.console.log(`Connecting to ${this.cameraName} using ICE`)
-        createStunResponder(this.audioSplitter)
+        if( rtpDescription.audio.port > 0 ) {
+          createStunResponder(this.audioSplitter)
+        }
+        if( rtpDescription.video.port > 0 ) {
+          createStunResponder(this.videoSplitter)
+        }
 
         sendStunRequests()
       } else {
@@ -111,7 +137,11 @@ export class SipSession extends Subscribed {
       }
 
       this.audioSplitter.once('message', () => {
-        this.console.log(`Audio stream latched for ${this.cameraName}`)
+        this.console.log(`Audio stream latched for ${this.cameraName}, port: ${this.rtpOptions.audio.port}`)
+      })
+
+      this.videoSplitter.once('message', () => {
+        this.console.log(`Video stream latched for ${this.cameraName}, port: ${this.rtpOptions.video.port}`)
       })
 
       return rtpDescription
@@ -123,7 +153,7 @@ export class SipSession extends Subscribed {
   }
 
   static async reserveRtpRtcpPorts() {
-    const ports = await reservePorts({ count: 2, type: 'udp' })
+    const ports = await reservePorts({ count: 4, type: 'udp' })
     return ports
   }
 
@@ -134,15 +164,19 @@ export class SipSession extends Subscribed {
     this.hasCallEnded = true
 
     if (sendBye) {
-      await this.sipCall.sendBye().catch(this.console.log)
+      await this.sipCall.sendBye().catch(this.console.error)
     }
 
     // clean up
+    this.console.log("sip-session callEnded")
     this.onCallEndedSubject.next(null)
     this.sipCall.destroy()
     this.audioSplitter.close()
     this.audioRtcpSplitter.close()
+    this.videoSplitter.close()
+    this.videoRtcpSplitter.close()
     this.unsubscribe()
+    this.console.log("sip-session callEnded: done")
   }
 
   async stop() {