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Incoming call, no incoming audio

Rich_Rodecker
New Here ,
Dec 10, 2010 Dec 10, 2010

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I think this is an IP address issue again.  I'm able to successfully make an incoming call to the flash phone instance, but I cannot receive incoming audio, and after a few seconds the call is ended. I think i just need to figure out how to set the globalAddress for the default profile.

Here is a dump of the log for the call:

2010-12-10::16:56:11.679 DEBUG SIP 1584 Incoming call-leg 3072b28 was created

2010-12-10::16:56:11.679 DEBUG SIP 1584 Received Call msg type : -1

2010-12-10::16:56:11.679 DEBUG SIPLEG 1584 [LEG ID:0] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_OFFERING Reason=RVSIP_CALL_LEG_REASON_REMOTE_INVITING Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING

2010-12-10::16:56:11.679 DEBUG SIP 1584 Trying to find matching sip profile in Sip Database

2010-12-10::16:56:11.679 DEBUG SIP 1584 Trying to find matching sip profile through TO hdr for username : 1000

2010-12-10::16:56:11.679 DEBUG SIP 1584 Trying to find matching sip profile through FROM hdr for username : 3109238566

2010-12-10::16:56:11.679 DEBUG SIP 1584 Trying to find matching sip profile through CONTACT hdr for username : 3109238566

2010-12-10::16:56:11.679 DEBUG SIP 1584 No matching profile found - using the default profile now

2010-12-10::16:56:11.679 DEBUG SIP 1584 getMGAddressToUse::localIp not set, Using MG Address=10.161.23.77

2010-12-10::16:56:11.680 DEBUG SIP 1584 Setting connection address to 10.161.23.77

2010-12-10::16:56:11.680 DEBUG SIP 1584 No Video session added in SDP

2010-12-10::16:56:11.680 DEBUG SIP 1584 -- Some fields from offer SDP message: --

2010-12-10::16:56:11.680 DEBUG SIP 1584 version field     : 0

2010-12-10::16:56:11.680 DEBUG SIP 1584 session           : Flash Media Gateway 2,0,0,72

2010-12-10::16:56:11.680 DEBUG SIP 1584 origin  field:

2010-12-10::16:56:11.680 DEBUG SIP 1584 username         : MG

2010-12-10::16:56:11.680 DEBUG SIP 1584 session id       : 2630303

2010-12-10::16:56:11.680 DEBUG SIP 1584 version          : 2630303

2010-12-10::16:56:11.680 DEBUG SIP 1584 address          : 10.161.23.77

2010-12-10::16:56:11.680 DEBUG SIP 1584 connection field:

2010-12-10::16:56:11.680 DEBUG SIP 1584 nettype enum     : 1

2010-12-10::16:56:11.680 DEBUG SIP 1584 addr type enum   : 1

2010-12-10::16:56:11.680 DEBUG SIP 1584 address          : 10.161.23.77

2010-12-10::16:56:11.680 DEBUG SIP 1584 Session Time 0:

2010-12-10::16:56:11.680 DEBUG SIP 1584 start            : 0

2010-12-10::16:56:11.680 DEBUG SIP 1584 end              : 0

2010-12-10::16:56:11.680 DEBUG SIP 1584 Media Descr 0:

2010-12-10::16:56:11.680 DEBUG SIP 1584 media type enum  : 1

2010-12-10::16:56:11.680 DEBUG SIP 1584 port             : 6000

2010-12-10::16:56:11.680 DEBUG SIP 1584 protocol enum    : 1

2010-12-10::16:56:11.680 DEBUG SIP 1584 format 0         : 0

2010-12-10::16:56:11.680 DEBUG SIP 1584 format 1         : 111

2010-12-10::16:56:11.680 DEBUG SIP 1584 format 2         : 101

2010-12-10::16:56:11.681 DEBUG SIP 1584 codec PCMU was approved, payload type : 0

2010-12-10::16:56:11.681 DEBUG SIP 1584 codec telephone-event was approved, payload type : 101

2010-12-10::16:56:11.685 DEBUG CALLLEG 1584 [LEG ID:5] - Allocated new Call Leg(sip)

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 0

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - State ALLOC

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - State Change ALLOC -> START

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 1

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Call Leg START

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - State Change START -> RING

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 2

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Call Leg RINGing

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - State Change RING -> SENDRECV

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 3

2010-12-10::16:56:11.687 DEBUG SIPLEG 3184 [LEG ID:5] - sendrecvHandler called...

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Call Leg SENDRECV

2010-12-10::16:56:11.687 DEBUG CORE 3184 [LEG ID:5] - Processing <Context:rtmp>

2010-12-10::16:56:11.687 DEBUG CORE 3184 [LEG ID:5] - Testing condition <var:destNum, varData:1000> to <value:^1...$>

2010-12-10::16:56:11.687 DEBUG CORE 3184 num matches: 1

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - State Change SENDRECV -> EXEC

2010-12-10::16:56:11.687 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 4

2010-12-10::16:56:11.688 DEBUG SIPLEG 3184 [LEG ID:5] - executeHandler called...

2010-12-10::16:56:11.688 DEBUG CALLLEG 3184 [LEG ID:5] - Call Leg EXECUTE

2010-12-10::16:56:11.688 DEBUG CORE 3184 [LEG ID:5] - Processing <Context:rtmp>

2010-12-10::16:56:11.688 DEBUG CORE 3184 [LEG ID:5] - Testing condition <var:destNum, varData:1000> to <value:^1...$>

2010-12-10::16:56:11.688 DEBUG CORE 3184 num matches: 1

2010-12-10::16:56:11.688 DEBUG CALLLEG 3184 [LEG ID:5] - Execute bridge(rtmp|${destNum}@tff)

2010-12-10::16:56:11.688 DEBUG CALLLEG 3184 [LEG ID:5] - Expanded String bridge(rtmp|1000@tff) (ID:1)

2010-12-10::16:56:11.688 DEBUG SIPLEG 3184 [LEG ID:5] - Call in progress

2010-12-10::16:56:11.688 INFO CALLLEG 3184 [LEG ID:5] - Ringing

2010-12-10::16:56:11.688 DEBUG CALLLEG 3184 [LEG ID:6] - Allocated new Call Leg(rtmp)

2010-12-10::16:56:11.688 INFO RTMPLEG 3184 [LEG ID:6] - setting silence level to 6 for Leg

2010-12-10::16:56:11.688 INFO RTMPLEG 3184 [LEG ID:6] - setting silence Timeout to 500 for Leg

2010-12-10::16:56:11.692 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 0

2010-12-10::16:56:11.692 DEBUG CALLLEG 1716 [LEG ID:6] - State ALLOC

2010-12-10::16:56:11.694 DEBUG CALLLEG 1716 [LEG ID:6] - State Change ALLOC -> START

2010-12-10::16:56:11.694 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 1

2010-12-10::16:56:11.694 DEBUG RTMPLEG 1716 [LEG ID:6] - Inside start handler, creating stream

2010-12-10::16:56:11.694 DEBUG CALLLEG 1716 [LEG ID:6] - Call Leg START

2010-12-10::16:56:11.695 INFO RTMP 4748 Received onStatus <Success> code <status> classType <StreamCreated> description <> details

2010-12-10::16:56:11.698 INFO RTMP 4748 Received onStatus <NetStream.Publish.Start> code <status> classType <fmg/tff/3 is now published.> description <(null)> details

2010-12-10::16:56:11.953 INFO RTMP 4748 Received onStatus <Success> code <status> classType <StreamCreated> description <> details

2010-12-10::16:56:11.955 DEBUG CALLLEG 1716 [LEG ID:6] - State Change START -> RING

2010-12-10::16:56:11.955 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 2

2010-12-10::16:56:11.955 DEBUG CALLLEG 1716 [LEG ID:6] - Call Leg RINGing

2010-12-10::16:56:11.955 INFO RTMP 4748 Received onStatus <NetStream.Play.Reset> code <status> classType <Playing and resetting 1000.> description <(null)> details

2010-12-10::16:56:11.955 INFO RTMP 4748 Received onStatus <NetStream.Play.Start> code <status> classType <Started playing 1000.> description <(null)> details

2010-12-10::16:56:11.956 DEBUG SIPLEG 3184 [LEG ID:5] - Call in progress

2010-12-10::16:56:11.956 DEBUG CALLLEG 3184 [LEG ID:5] -  Variable Value Set (bridgeUUID=b81c32d9-c9fb-432b-808c-ec7f9d4e8163)

2010-12-10::16:56:11.956 DEBUG CALLLEG 3184 [LEG ID:6] -  Variable Value Set (bridgeUUID=a0313019-2d5e-4aae-ab34-a06acb804338)

2010-12-10::16:56:11.956 DEBUG SIPLEG 3184 [LEG ID:5] - Call ringing

2010-12-10::16:56:11.956 DEBUG CALLLEG 1716 [LEG ID:6] - State Change RING -> HOLD

2010-12-10::16:56:11.956 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 5

2010-12-10::16:56:11.956 DEBUG CALLLEG 1716 [LEG ID:6] - Call Leg HOLD

2010-12-10::16:56:12.001 INFO RTMP 4748 Received onStatus <NetStream.Play.PublishNotify> code <status> classType <1000 is now published.> description <(null)> details

2010-12-10::16:56:16.371 INFO CALLLEG 2568 [LEG ID:6] - Leg has been answered

2010-12-10::16:56:16.371 DEBUG CALLLEG 1716 [LEG ID:6] - State Change HOLD -> SENDRECV

2010-12-10::16:56:16.371 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 3

2010-12-10::16:56:16.372 DEBUG SIPLEG 3184 [LEG ID:5] - Call answered

2010-12-10::16:56:16.372 DEBUG SIP 3184 Port Allocated for Media transfer : 10006

2010-12-10::16:56:16.372 DEBUG SIP 3184 [LEG ID:5] - RTP Buffer size set to : 8192

2010-12-10::16:56:16.372 DEBUG SIP 3184 [LEG ID:5] - RTP session created successfully - localAddr - 10.161.23.77, localPort - 10006  remoteAddr - 216.52.221.141, remotePort - 16826

2010-12-10::16:56:16.373 DEBUG SIPLEG 3184 [LEG ID:5] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_ACCEPTED Reason=RVSIP_CALL_LEG_REASON_LOCAL_ACCEPTED Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING

2010-12-10::16:56:16.373 INFO CALLLEG 3184 [LEG ID:5] - Leg has been answered

2010-12-10::16:56:16.433 DEBUG CALLLEG 1716 [LEG ID:5] - Write Buffer at 320 bytes to adjust 512->320

2010-12-10::16:56:48.387 DEBUG SIPLEG 1584 [LEG ID:5] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_DISCONNECTING Reason=RVSIP_CALL_LEG_REASON_LOCAL_DISCONNECTING Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING

2010-12-10::16:56:48.414 DEBUG SIP 1584 Received Call msg type : 200

2010-12-10::16:56:48.415 DEBUG SIPLEG 1584 [LEG ID:5] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_DISCONNECTED Reason=RVSIP_CALL_LEG_REASON_DISCONNECTED Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING

2010-12-10::16:56:48.415 DEBUG SIP 1584 Callleg disconnect cause is 200

2010-12-10::16:56:48.415 DEBUG CALLLEG 1584 [LEG ID:5] - State Change EXEC -> HANGUP

2010-12-10::16:56:48.415 DEBUG SIPLEG 1584 [LEG ID:0] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_TERMINATED Reason=RVSIP_CALL_LEG_REASON_CALL_TERMINATED Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING

2010-12-10::16:56:48.415 DEBUG SIP 1584 [LEG ID:0] - Callleg disconnect cause was undefined, now set to 500 for reason RVSIP_CALL_LEG_REASON_CALL_TERMINATED

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:5] - Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:6] - Bridging Completed for

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:6] - Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:960

2010-12-10::16:56:48.421 INFO CALLLEG 1716 [LEG ID:6] - Hangup [SENDRECV] [OK]

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:6] - Going For State 7

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:6] - Call Leg HANGUP, cause: OK

2010-12-10::16:56:48.421 INFO CALLLEG 1716 [LEG ID:6] - CallLeg Ended

2010-12-10::16:56:48.421 DEBUG CALLLEG 1716 [LEG ID:6] - Cleaning up Leg [HANGUP]

2010-12-10::16:56:48.423 INFO RTMP 4748 Received onStatus <NetStream.Unpublish.Success> code <status> classType <fmg/tff/3 is now unpublished.> description <(null)> details

2010-12-10::16:56:48.423 INFO RTMP 4748 Received onStatus <NetStream.Play.Stop> code <status> classType <Stopped playing 1000.> description <(null)> details

2010-12-10::16:56:48.424 DEBUG CALLLEG 3184 [LEG ID:6] - Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1153

2010-12-10::16:56:48.424 DEBUG CALLLEG 3184 [LEG ID:5] - Bridging Completed for

2010-12-10::16:56:48.424 DEBUG CALLLEG 3184 [LEG ID:5] - Hangup Call (cause 200), from FMSMGAppNodesHelper.cpp:1251

2010-12-10::16:56:49.424 DEBUG CALLLEG 3184 [LEG ID:5] - App  bridge(rtmp|1000@tff) Returns 0 (next ID:1)

2010-12-10::16:56:49.424 DEBUG CALLLEG 3184 [LEG ID:5] - Going For State 7

2010-12-10::16:56:49.424 DEBUG SIPLEG 3184 [LEG ID:5] - hangupHandler called...default

2010-12-10::16:56:49.424 DEBUG SIP 3184 Closing audio RTP session

2010-12-10::16:56:49.424 DEBUG SIP 3184 closed audio socket

2010-12-10::16:56:49.424 DEBUG SIP 3184 closed video socket

2010-12-10::16:56:49.424 DEBUG SIP 3184 Removing Call Leg Entry for UUID a0313019-2d5e-4aae-ab34-a06acb804338

2010-12-10::16:56:49.424 DEBUG SIP 3184 call reject failed, will try disconnecting it

2010-12-10::16:56:49.434 DEBUG SIP 3184 Disconnected call name : default

2010-12-10::16:56:49.434 DEBUG CALLLEG 3184 [LEG ID:5] - Call Leg HANGUP, cause: OK

2010-12-10::16:56:49.434 INFO CALLLEG 3184 [LEG ID:5] - CallLeg Ended

2010-12-10::16:56:49.434 DEBUG CALLLEG 3184 [LEG ID:5] - Cleaning up Leg [HANGUP]

2010-12-10::16:56:49.434 DEBUG SIPLEG 3184 [LEG ID:5] - deleting sip leg

2010-12-10::16:56:49.434 DEBUG SIP 3184 Closing audio RTP session

2010-12-10::16:56:49.434 DEBUG SIP 3184 closed audio socket

2010-12-10::16:56:49.434 DEBUG SIP 3184 closed video socket

2010-12-10::16:56:50.421 DEBUG RTMP 1716 [min-size, count]: [0,0],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,0], [9,0], [>=10,0]

2010-12-10::16:56:50.421 DEBUG RTMP 1716 [max-size, count]: [0,0],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,0], [9,0], [>=10,0]

2010-12-10::16:56:50.421 DEBUG RTMP 1716 [variation, count]: [0,0],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,0], [9,0], [>=10,0]

2010-12-10::16:56:50.421 DEBUG RTMP 1716 [min-size, count]: [0,2],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,0], [9,0], [>=10,0]

2010-12-10::16:56:50.422 DEBUG RTMP 1716 [max-size, count]: [0,0],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,1], [9,0], [>=10,1]

2010-12-10::16:56:50.422 DEBUG RTMP 1716 [variation, count]: [0,0],[1,0], [2,0], [3,0], [4,0], [5,0], [6,0], [7,0], [8,1], [9,0], [>=10,1]

my entire workflow.xml is this:

<WorkFlow>

<ContextList>

<Context name="sipGatewayContext">

<!-- call rtmp phone -->

<Condition variable="destNum" value="^.*$">

<AppNode sequence="1" app="bridge" args="rtmp|${destNum}@tff"/>

<AppNode sequence="2" app="hangup" args="null"/>

</Condition>

</Context>

<Context name="rtmp">

<!-- play moh.raw file and then hangup -->

<!-- call rtmp phone -->

<Condition variable="destNum" value="^1...$">

<AppNode sequence="1" app="bridge" args="rtmp|${destNum}@tff"/>

<AppNode sequence="2" app="hangup" args="null"/>

</Condition>

<!-- call made on SIP g/w with the destination number -->

<Condition variable="destNum" value="^.*$">

<AppNode sequence="1" app="bridge" args="sip|${destNum}@sipGateway"/>

<AppNode sequence="2" app="hangup" args="null"/>

</Condition>

</Context>

</ContextList>

</WorkFlow>

and I have 2 profiles in rtmp.xml:

<Profiles>

<!-- To enable using auto profileID if entry not found in any <Profile> tag below. -->

<!-- Default value is false -->

<EnableAutoProfile>true</EnableAutoProfile>

<!--  DefaultContext value will be used for all AutoProfiles. Default value is default  -->

<DefaultContext>rtmp</DefaultContext>

<!-- The default passcode that FMG will use for all FMS apps for which no specific -->

<!-- is mentioned in xml configurations -->

<DefaultPassCode>passcode_default</DefaultPassCode>

<!-- Specifying list of RTMP <profile>, all fields are required, please refer -->

<!-- to configuration guide to know more about restrictions. -->

<Profile>

    <!-- Unique profile ID accross rtmp plugin, This profileID can be used -->

<!-- in workflow.xml to specify call flows. -->

<ProfileID> profile_default </ProfileID>

<!-- FMS-Host, application & instance name are mandatory to specify an -->

<!-- application to which this profile ID is assigned -->

<Server> 184.72.XX.XXX </Server>

<Application> tff </Application>

<Instance> _definst_ </Instance>

<!-- the shared Passcode that FMG will use while connecting to this application instance -->

<!-- -->

<PassCode>passcode_default</PassCode>

<!-- Name of the workflow  context (in workflow.xml or workflow script -->

<!-- in FMG) which will be applied to the calls originating from this profileID -->

<Context> rtmp </Context>

</Profile>

<Profile>

    <!-- Unique profile ID accross rtmp plugin, This profileID can be used -->

<!-- in workflow.xml to specify call flows. -->

<ProfileID> tff </ProfileID>

<!-- FMS-Host, application & instance name are mandatory to specify an -->

<!-- application to which this profile ID is assigned -->

<Server> 184.72.XX.XXX </Server>

<Application> tff </Application>

<Instance> _definst_ </Instance>

<!-- the shared Passcode that FMG will use while connecting to this application instance -->

<!-- -->

<PassCode>passcode_default</PassCode>

<!-- Name of the workflow  context (in workflow.xml or workflow script -->

<!-- in FMG) which will be applied to the calls originating from this profileID -->

<Context> rtmp </Context>

</Profile>

</Profiles>

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replies 16 Replies 16
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Piyush-G
Explorer ,
Dec 12, 2010 Dec 12, 2010

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There are two alternative ways to specify a default profile -

- Set <EnableAutoProfile> tag to true and <DefaultContext> to a workflow context.

- Configure a profile whose <profileID> is default. When default profileID is specified, then it takes precedence over <EnableAutoProfile> and <DefaultContext>

I think here we need to use the second way in order to set the <globalAddress> for default profile.

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Rich_Rodecker
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Dec 14, 2010 Dec 14, 2010

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In Response To Piyush-G

Still having problems with this.  I set up a default profile in sip.xml:

<Profile>

<!-- Unique name for the profile. This ID is used in workflow to identify where the call will be placed -->

<profileID> default </profileID>

<!-- default SIP IP/hostname, in case the remote SIP client is not meant to register,

to bind to dynamic address use '*' in which case the sip phone would have to register at FMG -->

<!--<remoteSipHost> * </remoteSipHost-->

<remoteSipHost> sip.xxx.com:5060 </remoteSipHost>

<!-- Global Address if the FMG server is on a local network and need to connect to outside network -->

<globalAddress> 184.XX.XX.XXX </globalAddress>

<!-- Context in which incoming calls will be handled, this points to the context in workflow.xml -->

<context> rtmp </context>

<!-- supported Codecs for this SIP profile

All the codecs here should be from the CodecList defined earlier,

If no codec is specified then codecs specified in CodecList would be used

DTMF is always supported -->

<supportedCodecs>

<codecID> G711u </codecID>

</supportedCodecs>

</Profile>

In the log I'm seeing this line:

2010-12-14::16:03:02.560 DEBUG SIP 5668 getMGAddressToUse::localIp not set, Using MG Address=10.161.23.77

I'm guessing that might have something to do with it...where can I force it to use the external IP address?

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Rich_Rodecker
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Dec 14, 2010 Dec 14, 2010

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In Response To Rich_Rodecker

should I be setting the external IP in conf/_defaultRoot_/Adaptor.xml?  

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Rich_Rodecker
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Dec 15, 2010 Dec 15, 2010

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In Response To Rich_Rodecker

Ok, I have been going at this for the entire day and I still can't figure out what's going on.

I set up my default profile in sip.xml:

          <Profile>
               <!-- Unique name for the profile. This ID is used in workflow to identify where the call will be placed -->
               <profileID> default </profileID>
               <!-- Global Address if the FMG server is on a local network and need to connect to outside network -->
               <globalAddress> 184.72.XX.XXX </globalAddress>
               <remoteSipHost> sip.xxx.com</remoteSipHost>
               <!-- Context in which incoming calls will be handled, this points to the context in workflow.xml -->
               <context> rtmp </context>
               <!-- supported Codecs for this SIP profile
               All the codecs here should be from the CodecList defined earlier,
               If no codec is specified then codecs specified in CodecList would be used
               DTMF is always supported -->
               <supportedCodecs>
                    <codecID> G711u </codecID>
                    <codecID> speex </codecID> 
                    <codecID> H264 </codecID>
                    </supportedCodecs>
          </Profile>

and when FMG starts I can see this in the logs:

2010-12-15::00:52:33.303 DEBUG SIP 4080 Setting Global Address for profile  to 184.72.34.176

so I'm guessing that's the default profile even though it doesn't show the name.  Then when the incoming call starts, I see this in the logs:

2010-12-15::00:53:16.574     DEBUG     SIP     4080     getMGAddressToUse::localIp not set, Using MG Address=10.161.23.77

2010-12-15::00:53:16.574     DEBUG     SIP     4080      Setting connection address to 10.161.23.77

2010-12-15::00:53:16.574     DEBUG     SIP     4080     -- Some fields from offer SDP message: --

2010-12-15::00:53:16.575     DEBUG     SIP     4080     version field     : 0

2010-12-15::00:53:16.575     DEBUG     SIP     4080     session           : Flash Media Gateway 2,0,0,72

2010-12-15::00:53:16.575     DEBUG     SIP     4080     origin  field:

2010-12-15::00:53:16.575     DEBUG     SIP     4080      username         : MG

2010-12-15::00:53:16.575     DEBUG     SIP     4080      session id       : 2630303

2010-12-15::00:53:16.575     DEBUG     SIP     4080      version          : 2630303

2010-12-15::00:53:16.575     DEBUG     SIP     4080      address          : 10.161.23.77

2010-12-15::00:53:16.575     DEBUG     SIP     4080     connection field:

2010-12-15::00:53:16.575     DEBUG     SIP     4080      nettype enum     : 1

2010-12-15::00:53:16.575     DEBUG     SIP     4080      addr type enum   : 1

2010-12-15::00:53:16.575     DEBUG     SIP     4080      address          : 10.161.23.77

2010-12-15::00:53:16.575     DEBUG     SIP     4080     Session Time 0:

2010-12-15::00:53:16.575     DEBUG     SIP     4080      start            : 0

2010-12-15::00:53:16.575     DEBUG     SIP     4080      end              : 0

2010-12-15::00:53:16.575     DEBUG     SIP     4080     Media Descr 0:

2010-12-15::00:53:16.575     DEBUG     SIP     4080      media type enum  : 1

2010-12-15::00:53:16.575     DEBUG     SIP     4080      port             : 6000

2010-12-15::00:53:16.575     DEBUG     SIP     4080      protocol enum    : 1

2010-12-15::00:53:16.575     DEBUG     SIP     4080      format 0         : 0

2010-12-15::00:53:16.575     DEBUG     SIP     4080      format 1         : 111

2010-12-15::00:53:16.575     DEBUG     SIP     4080      format 2         : 101

2010-12-15::00:53:16.575     DEBUG     SIP     4080     Media Descr 1:

2010-12-15::00:53:16.575     DEBUG     SIP     4080      media type enum  : 3

2010-12-15::00:53:16.575     DEBUG     SIP     4080      port             : 6000

2010-12-15::00:53:16.575     DEBUG     SIP     4080      protocol enum    : 1

2010-12-15::00:53:16.575     DEBUG     SIP     4080      format 0         : 99

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Guest
Dec 15, 2010 Dec 15, 2010

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Can you run wireshark on the machine where FMG is running and capture the SIP (port 5060) packets and RTP/UDP data and post the file. Will help in diagnosis.

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Rich_Rodecker
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Dec 15, 2010 Dec 15, 2010

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In Response To Deleted User

I downloaded wiresharek but i'm nto sure how to capture, can you point me in the right direction?

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Guest
Dec 15, 2010 Dec 15, 2010

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In Response To Rich_Rodecker

Install

Run

Capture->Interfaces -> Click "Start" for the interface used for SIP (10.161.23.77 i guess)

This will start the capture

write "udp.port==5060" in the Filter box and press "Apply" . now you will only see the SIP messages.

trying making a call now.

one you are done with all the experiments do

Capture->Stop

File -> Save As -> select the "Displayed" and save the file.

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Rich_Rodecker
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Dec 15, 2010 Dec 15, 2010

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Here is a link to the dump, let me know if this works for you:  http://www.f1fd.com/calldump.pcap

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Piyush-G
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Dec 15, 2010 Dec 15, 2010

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In Response To Rich_Rodecker

Thanks for providing the info. The external IP address NATting isn't working properly with default profile. For a workaround, need to use a non-default profile.

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Rich_Rodecker
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Dec 15, 2010 Dec 15, 2010

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Ok, if I delete the default SIP profile and add one for user 1000:

               <!-- Unique name for the profile. This ID is used in workflow to identify where the call will be placed -->
               <profileID> 1000 </profileID>
               <userName> 1000 </userName> 
               <password></password>
               <!-- Global Address if the FMG server is on a local network and need to connect to outside network -->
               <globalAddress> 184.72.XX.XXX </globalAddress>
               <remoteSipHost> sip.xxx.com:5060 </remoteSipHost>
               <doRegister>0</doRegister>               
               <!-- Context in which incoming calls will be handled, this points to the context in workflow.xml -->
               <context> rtmp </context>
               <!-- supported Codecs for this SIP profile
               All the codecs here should be from the CodecList defined earlier,
               If no codec is specified then codecs specified in CodecList would be used
               DTMF is always supported -->
               <supportedCodecs>
                    <codecID> speex </codecID> 
                    </supportedCodecs>
          </Profile>
I get this in the log:
2010-12-15::12:44:06.167     DEBUG     SIPLEG     5928     [LEG ID:0] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_OFFERING Reason=RVSIP_CALL_LEG_REASON_REMOTE_INVITING Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING
2010-12-15::12:44:06.167     DEBUG     SIP     5928     Trying to find matching sip profile in Sip Database
2010-12-15::12:44:06.167     DEBUG     SIP     5928     Trying to find matching sip profile through TO hdr for username : 1000
2010-12-15::12:44:06.167     DEBUG     SIP     5928     Trying to find matching sip profile through FROM hdr for username : 3109238566
2010-12-15::12:44:06.167     DEBUG     SIP     5928     Trying to find matching sip profile through CONTACT hdr for username : 3109238566
2010-12-15::12:44:06.168     DEBUG     SIP     5928     Local Interface : 10.161.23.77 will be used for dest ip : 216.52.221.141
2010-12-15::12:44:06.168     DEBUG     SIP     5928     No matching profile found - using the default profile now
2010-12-15::12:44:06.169     WARNING     SIP     5928     Call Rejected : Reason : Error : Call Could not be created : Reason : default profile not found (internal error)
2010-12-15::12:44:06.170     DEBUG     SIPLEG     5928     [LEG ID:0] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_DISCONNECTED Reason=RVSIP_CALL_LEG_REASON_LOCAL_REJECT Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING
2010-12-15::12:44:06.170     DEBUG     SIP     5928     [LEG ID:0] - Callleg disconnect cause was undefined, now set to 500 for reason RVSIP_CALL_LEG_REASON_LOCAL_REJECT
2010-12-15::12:44:06.191     DEBUG     SIP     5928     Received Call msg type : -1
2010-12-15::12:44:06.192     DEBUG     SIPLEG     5928     [LEG ID:0] - **callLegStateChangedEvHandler** State=RVSIP_CALL_LEG_STATE_TERMINATED Reason=RVSIP_CALL_LEG_REASON_CALL_TERMINATED Dir=RVSIP_CALL_LEG_DIRECTION_INCOMING
2010-12-15::12:44:06.192     DEBUG     SIP     5928     [LEG ID:0] - Callleg disconnect cause was undefined, now set to 500 for reason RVSIP_CALL_LEG_REASON_CALL_TERMINATED
I've tried various different settings for the "1000" profile and I always get the error about the default profile not found.
All I am trying to accomplish is to have an incoming sip call to 1000@fmg.ip.address routed to an instance of flashphone with an ID of 1000 running in the browser.  If i use a default profile, the call connection is made and the call starts, but I can't get incoming audio.  If I try to define a profile with username of 1000 then we get the errors above.

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hey guys, I gave up trying to get it running on E2, installed on another local windows computer and got both incoming and outgoing calls working.  Thanks for all the help...hopefully we'll be able to get it running on Amazon soon.

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hello Rich, this is ankush
I am able to call from Browser to phone but not vice versa. Please  suggest me wat can be the problem, also I want to know that what would  be the "Global Address". Is it local IP address or Sip service providers  ip ?
PLease answer this and give example of a global address..
Thanks and Regards
Ankush
Message was edited by: ankushbagga

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hello Piyush,
I want to know that what would be the "Global Address". Is it local IP address or Sip service providers ip ?
PLease answer this and give example of a global address..
Thanks and Regards
Ankush

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hello Ankush,
   GlobalAddress would be needed only when the machine, running FMG, has local (internal) IP address and FMG needs to communicate with a SIP entity in the external network. In that case, GlobalAddress would be set to the external (or public) IP address to which IP address of the FMG machine is NATted. One way to find external IP address is open the URL http://www.whatismyip.com/ from that machine.
Regards,
Piyush

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hi piyush,
Thanks Piyush for the fast reply, i got my Global address from
the link you have posted, its 202.69.34.
I tried this in global Address tag of sipGateway profile in sip.xml
Still calling from phone to browser is not possible.
Please tell me wat i am missing
Regards
Ankush
 
 

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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Hi Ankush,
First of all ensure that the FMG machine is getting the SIP INVITE message - can be verified by running wireshark on FMG machine. If it is not, then one possibility could be that it is blocked at a firewall somewhere between FMG and the SIP service provider.
Regards,
Piyush

					
				
			
			
				
			
			
			
			
			
			
			
		

		
		
	

	
	


	
				       
    


		
	
	


		

	

		

			

	
		
			
        




              

            
          

            

                
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