Open gprada opened 11 years ago
I probably have the same issue, when openvpn is running. It works when i disable openvpn or remove the required iptable entry (to masquerade some packets).
" reason: EndedBySecurityDenial"
I suspect this means your upstream sip provider is ending the call. If this is resolved by disabling a VPN i suspect that some rewriting of the SIP from header is happening over the VPN :)
@ml1nk Could you please specify what you mean by: "or remove the required iptable entry (to masquerade some packets)." When shutting down openvpn, sipcmd works fine, but I need openVPN. So I would like to check out your workaround mentioned above. Thanks
@NCO3 If I remember correctly it wasn't a workaround at all. The iptable entries are used from openVPN, without them there is no proper connection. I couldn't solve the problem, so I installed openVPN on another server without sipcmd.
@ml1nk thanks for your response. I found a workaround: I just stop the openvpn by my openHAB home automation, let it do the call and activate openvpn again. Works perfectly! :-)
I also ran into this error and got a solution: When you have some/any virtual network interface(s) it will lead to this error.
I checked with ifconfig
and after i.e. ifconfig docker0 down
it worked.
After analyzing the debug opal logs with the -o
option of sipcmd I think the root cause is, that sipcmd is sending requests with all available interface IPs to the sip server which then will cause "EndedBySecurityDenial". This will also happen if you specify the ip with the -l
param (local address to listen on)
Get 2 errors on trying to call with sip cmd:
root@ubuntu:/usr/src/sipcmd-master# ./sipcmd -P sip -u 599 -c 1234 -o opal.log -x "c503;ws3000;d123;w200;h" -w 192.168.4.203 Starting sipcmd in debug mode Manager Init initialising SIP endpoint... TestChanAudio TestChanAudio Listening for SIP signalling on 0.0.0.0:5060 Assertion fail: Function pthread_setschedparam failed, file ptlib/unix/tlibthrd.cxx, line 757, Error=1
bort,ore dump, hrow exception, gnore? i
Ignoring. SIP listener up registered as sip:599@192.168.4.203 Created LocalEndPoint Main
Call
TestPhone::Main: calling "503" using gateway "192.168.4.203" at Fri Oct 4 11:19:05 2013
Setting up a call to: sip:503@192.168.4.203 LocalEndpoint::MakeConnection LocalEndpointCreateConnection LocalConnection OnIncomingConnection: token=L4410ea3e2 connection set up to sip:503@192.168.4.203 OnReleased: reason: EndedBySecurityDenial
Wait: waiting for 3000ms
OnReleased: reason: EndedBySecurityDenial OnClearedCall ~LocalConnection Wait: wait done
DTMF "123"
no call found with token=C49c691cd1 Problem running command sequence ("c503;ws3000;d123;w200;h"):
TestPhone::Main: shutting down TestPhone::Main: exiting... Exiting...
~Manager
Follow the opal log.
0:00.043 sipcmd Version 1.0.1 by Command line VoIP testphone on Unix Linux (2.6.32-22-pve-x86_64) with PTLib (v2.10.4 (svn:26606)) at 2013/10/4 11:19:02.312 0:00.043 sipcmd OpalMan Attached endpoint with prefix sip 0:00.043 sipcmd OpalEP Created endpoint: sip 0:00.044 sipcmd PWLib File handle high water mark set: 8 PUDPSocket 0:00.044 Opal Garbage:0x8b36700 PTLib Started thread 0x88a340 (20072) Opal Garbage:0x8b36700 0:00.044 sipcmd IfaceMon Initial interface list: 127.0.0.1 <00-00-00-00-00-00> (lo) 127.0.0.2 <00-00-00-00-00-00> (venet0) 192.168.4.208 <00-00-00-00-00-00> (venet0:0) ::1 <00-00-00-00-00-00> (lo)
0:00.044 sipcmd PTLIB Opened NetLink socket 0:00.044 sipcmd PWLib File handle high water mark set: 14 Thread unblock pipe 0:00.044 sipcmd PTLib Created thread 0x88eae0 0:00.044 sipcmd PTLib Thread high water mark set: 3 0:00.044 sipcmd PWLib File handle high water mark set: 16 Thread unblock pipe 0:00.044 sipcmd PTLib Created thread 0x88dae0 Housekeeper 0:00.044 sipcmd PTLib Thread high water mark set: 4 0:00.045 Housekeeper:0x8ab4700 PTLib Started thread 0x88dae0 (20074) Housekeeper:0x8ab4700 0:00.045 sipcmd OpalMan Attached endpoint with prefix sips 0:00.045 Network In...:0x8af5700 PTLib Started thread 0x88eae0 (20073) Network Interface Monitor:0x8af5700 0:00.045 sipcmd SIP Created endpoint. 0:00.045 sipcmd OpalMan Added route "local:.=sip:"
0:00.045 sipcmd OpalMan Added route "sip:. =local:"
0:00.045 sipcmd PWLib File handle high water mark set: 17 PUDPSocket
0:00.045 sipcmd MonSock Created socket bundle for all interfaces.
0:00.045 sipcmd PWLib File handle high water mark set: 18 PUDPSocket
0:00.045 sipcmd MonSock Created bundled UDP socket 192.168.4.208:5060
0:00.045 sipcmd PWLib File handle high water mark set: 20 Thread unblock pipe
0:00.045 sipcmd PTLib Created thread 0x894740 Opal Listener
0:00.046 sipcmd PTLib Thread high water mark set: 5
0:00.045 Network In...:0x8af5700 IfaceMon Started interface monitor thread.
0:00.046 sipcmd PWLib Assertion fail: Function pthreadsetschedparam failed, file ptlib/unix/tlibthrd.cxx, line 757, Error=1
0:00.046 Opal Listener:0x8a73700 PTLib Started thread 0x894740 (20075) Opal Listener:0x8a73700
0:00.046 Opal Listener:0x8a73700 Listen Started listening thread on udp$:5060
0:03.672 sipcmd SIP Start REGISTER
aor=599
remote=192.168.4.203
local=
contact=
authID=
realm=
expire=0
restore=30
minRetry=default
maxRetry=default
compatibility=FullyCompliant
0:03.673 sipcmd PWLib File handle high water mark set: 21 PUDPSocket
0:03.673 sipcmd SIP Constructed REGISTER handler for sip:599@192.168.4.203
0:03.673 sipcmd SIP Executing state change to Subscribing for REGISTER handler, target=sip:599@192.168.4.203, id=a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
0:03.673 sipcmd SIP Changing REGISTER handler from Unavailable to Subscribing, target=sip:599@192.168.4.203, id=a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
0:03.674 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:03.674 sipcmd SIP Created transport udp$192.168.4.203:5060
0:03.674 sipcmd OpalUDP Started connect to 192.168.4.203:5060
0:03.674 sipcmd OpalUDP Writing to interface 0 - "192.168.4.208%venet0:0"
0:03.675 sipcmd OpalMan Listener interfaces: associated transport=udp$192.168.4.208:5060
udp$192.168.4.208:5060
0:03.675 sipcmd SIP Transaction created.
0:03.676 sipcmd SIP Transaction remote address is udp$192.168.4.203:5060
0:03.676 sipcmd SIP Sending PDU (554 bytes) to: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0
REGISTER sip:192.168.4.203 SIP/2.0
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bK8a9429fa-752b-e311-8d69-f9c4e96053ec;rport
User-Agent: sipcmd/1.0.1
From: sip:599@192.168.4.203;tag=524029fa-752b-e311-8d69-f9c4e96053ec
Call-ID: a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
Organization: Command line VoIP testphone
To: sip:599@192.168.4.203
Contact: sip:599@192.168.4.208:5060;q=1
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
Expires: 3600
Content-Length: 0
Max-Forwards: 70
0:03.676 sipcmd OpalUDP Setting interface to 192.168.4.208%venet0:0 0:03.676 sipcmd SIP Transaction timers set: retry=10.000, completion=16.000 0:03.676 sipcmd OpalMan Attached endpoint with prefix local 0:03.676 sipcmd OpalEP Created endpoint: local 0:03.676 sipcmd LocalEP Created endpoint. 0:03.677 sipcmd OpalMan Set up call from local:* to sip:503@192.168.4.203 0:03.677 sipcmd Call Created Call[C49c691cd1] 0:03.677 sipcmd OpalMan Set up connection to "local:" 0:03.677 sipcmd OpalCon Created connection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd LocalCon Created connection with token "L4410ea3e2"
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalMan OnIncoming connection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd Call GetOtherPartyConnection Call[C49c691cd1]-EP[L4410ea3e2]
0:03.677 sipcmd OpalMan Searching for route "local:root sip:503@192.168.4.203"
0:03.677 sipcmd OpalMan Matched regex "^local:. .$" ("local:.")
0:03.677 sipcmd OpalMan Set up connection to "sip:503@192.168.4.203"
0:03.677 sipcmd OpalCon Created connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.677 sipcmd RFC2833 Handler created
0:03.678 sipcmd RFC2833 Handler created
0:03.678 sipcmd SIP Created connection.
0:03.678 sipcmd LocalCon Outgoing call routed to sip:503@192.168.4.203 for Call[C49c691cd1]-EP[L4410ea3e2]
0:03.678 sipcmd Call OnSetUp Call[C49c691cd1]-EP[L4410ea3e2]
0:03.678 sipcmd SIP SetUpConnection: sip:503@192.168.4.203
0:03.678 sipcmd SIP Connecting to sip:503@192.168.4.203 via sip:503@192.168.4.203
0:03.678 sipcmd SIP Setting new transport for destination "sip:503@192.168.4.203"
0:03.678 sipcmd SIP Found registrar on domain 192.168.4.203, using interface 192.168.4.208%venet0:0
0:03.678 sipcmd Opal Illegal IP transport port/service: "tcp$192.168.4.208%venet0:0"
0:03.678 sipcmd OpalUDP Binding to interface: 0.0.0.0:5060
0:03.678 sipcmd SIP Created transport udp$192.168.4.203:5060
0:03.678 sipcmd OpalUDP Started connect to 192.168.4.203:5060
0:03.688 sipcmd Call GetMediaFormats for Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd SIP Local media formats set:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd SIP Remote media formats set:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.688 sipcmd OpalUDP Writing to interface 0 - "192.168.4.208%venet0:0"
0:03.688 sipcmd SIP Getting local URI from registeration: sip:599@192.168.4.203
0:03.688 sipcmd SIP Updating dialog tag from "" to "cee129fa-752b-e311-8d69-f9c4e96053ec"
0:03.688 sipcmd SIP Remote dialog address from target: sip:503@192.168.4.203
0:03.689 sipcmd SIP INVITE transaction id=z9hG4bKa2932bfa-752b-e311-8d69-f9c4e96053ec created.
0:03.689 sipcmd SIP Creating INVITE request
0:03.689 sipcmd SIP Offering all configured media:
G.722.2,GSM-AMR,GSM-06.10,G.726-40k,G.726-32k,G.726-24k,G.726-16k,G.711-uLaw-64k,G.711-ALaw-64k,H.261,G.722-64k,G.722.1-24k,G.722.1-32k,SILK-16,SpeexIETFWide-20.6k,SpeexWB,SpeexWide-20.6k,LPC-10,MS-GSM,MS-IMA-ADPCM,SILK-8,SpeexIETFNarrow-11k,SpeexIETFNarrow-15k,SpeexIETFNarrow-18.2k,SpeexIETFNarrow-24.6k,SpeexIETFNarrow-5.95k,SpeexIETFNarrow-8k,SpeexNB,SpeexWNarrow-8k,T.38,UserInput/RFC2833,NamedSignalEvent,H.261-CIF,H.261-QCIF,RFC4175_YCbCr-4:2:0,theora,MSRP,SIP-IM,T.140,H.224/H323AnnexQ
0:03.689 sipcmd SIP Offering media type audio in SDP
0:03.689 sipcmd Call IsMediaBypassPossible Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec] session 1
0:03.689 sipcmd OpalMan IsMediaBypassPossible: session 1
0:03.689 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:03.689 sipcmd RTP Cannot find media session 1
0:03.689 sipcmd RTP Cannot find RTP session 1
0:03.690 sipcmd PTLib Created PXConfig 0x8b2120
0:03.690 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:03.690 sipcmd RTP_UDP Session 1, created with NAT flag set to 0
0:03.690 sipcmd PWLib File handle high water mark set: 22 PUDPSocket
0:03.690 sipcmd PWLib File handle low water mark set: 21 PUDPSocket
0:03.690 sipcmd RTP_UDP Session 1 created: 192.168.4.208:5000-5001 ssrc=916538279
0:03.690 sipcmd RTP Creating new session RTP_UDP
0:03.690 sipcmd RTPEp Session 1, remembering local RTP port 5000 on connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.690 sipcmd RTP Found existing media session 1
0:03.691 sipcmd MediaFormat Validation of merge for media option "BitRate" failed.
0:03.691 sipcmd SDP SDP not including SpeexIETFWide-20.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexWide-20.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including MS-GSM as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including MS-IMA-ADPCM as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-11k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-15k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-18.2k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-24.6k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-5.95k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexIETFNarrow-8k as it is not a SIP transportable format
0:03.691 sipcmd SDP SDP not including SpeexWNarrow-8k as it is not a SIP transportable format
0:03.691 sipcmd MediaFormat Merging UserInput/RFC2833 into UserInput/RFC2833
0:03.691 sipcmd RFC2833 Set tx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:03.691 sipcmd RFC2833 Set rx pt=[pt=101], events="0-16,32,36" for UserInput/RFC2833
0:03.691 sipcmd MediaFormat Merging NamedSignalEvent into NamedSignalEvent
0:03.691 sipcmd RFC2833 Set tx pt=[pt=100], events="192-193" for NamedSignalEvent
0:03.691 sipcmd RFC2833 Set rx pt=[pt=100], events="192-193" for NamedSignalEvent
0:03.692 sipcmd SIP Offering media type video in SDP
0:03.692 sipcmd Call IsMediaBypassPossible Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec] session 2
0:03.692 sipcmd OpalMan IsMediaBypassPossible: session 2
0:03.692 sipcmd OpalCon IsMediaBypassPossible: default returns false
0:03.692 sipcmd RTP Cannot find media session 2
0:03.692 sipcmd RTP Cannot find RTP session 2
0:03.692 sipcmd VoIP Metrics RTCP_XR_Metrics created.
0:03.692 sipcmd RTP_UDP Session 2, created with NAT flag set to 0
0:03.692 sipcmd PWLib File handle high water mark set: 23 PUDPSocket
0:03.692 sipcmd PWLib File handle high water mark set: 24 PUDPSocket
0:03.692 sipcmd PWLib File handle low water mark set: 23 PUDPSocket
0:03.692 sipcmd RTP_UDP SetOption(23,8,1048576) failed, even though it said it succeeded!
0:03.692 sipcmd RTP_UDP SetOption(23,8,524288) failed, even though it said it succeeded!
0:03.692 sipcmd RTP_UDP SetOption(23,8,262144) succeeded.
0:03.692 sipcmd RTP_UDP Session 2 created: 192.168.4.208:5002-5003 ssrc=1440891105
0:03.692 sipcmd RTP Creating new session RTP_UDP
0:03.692 sipcmd RTPEp Session 2, remembering local RTP port 5002 on connection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.692 sipcmd RTP Found existing media session 2
0:03.692 sipcmd SDP SDP not including H.261-CIF as it is not a SIP transportable format
0:03.692 sipcmd SDP SDP not including H.261-QCIF as it is not a SIP transportable format
0:03.693 sipcmd SIP Transaction remote address is udp$192.168.4.203:5060
0:03.693 sipcmd OpalPlugin to_customised_options:
Format Name = SILK-16
Media Type = audio
Payload Type = [pt=103]
Encoding Name = SILK
Channels (R/W) = 1 UnsignedInt
Clock Rate (R/O) = 16000 UnsignedInt
Complexity (R/O) = 1 UnsignedInt
Frame Time (R/O) = 320 UnsignedInt
Max Bit Rate (R/O) = 30000 UnsignedInt
Max Frame Size (R/O) = 75 UnsignedInt
Max Frames Per Packet (R/O) = 5 UnsignedInt
Needs Jitter (R/O) = 1 Boolean
Protocol (R/O) = SIP String
Rx Frames Per Packet (R/W) = 5 UnsignedInt
Tx Frames Per Packet (R/W) = 2 UnsignedInt
Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean
Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean
0:03.694 sipcmd OpalPlugin to_customised_options: Format Name = SILK-8 Media Type = audio Payload Type = [pt=102] Encoding Name = SILK Channels (R/W) = 1 UnsignedInt Clock Rate (R/O) = 8000 UnsignedInt Complexity (R/O) = 1 UnsignedInt Frame Time (R/O) = 160 UnsignedInt Max Bit Rate (R/O) = 20000 UnsignedInt Max Frame Size (R/O) = 50 UnsignedInt Max Frames Per Packet (R/O) = 5 UnsignedInt Needs Jitter (R/O) = 1 Boolean Protocol (R/O) = SIP String Rx Frames Per Packet (R/W) = 5 UnsignedInt Tx Frames Per Packet (R/W) = 2 UnsignedInt Use DTX (R/O) = 0 FMTP name: usedtx (0) Boolean Use In-Band FEC (R/O) = 1 FMTP name: useinbandfec (1) Boolean
0:03.695 sipcmd OpalPlugin to_customised_options: Format Name = H.261 Media Type = video Payload Type = H261 Encoding Name = h261 Annex D (R/O) = 0 FMTP name: D (0) Boolean CIF MPI (R/W) = 1 FMTP name: CIF (33) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 621700 UnsignedInt Max Rx Frame Height (R/O) = 288 UnsignedInt Max Rx Frame Width (R/O) = 352 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt Min Rx Frame Height (R/O) = 144 UnsignedInt Min Rx Frame Width (R/O) = 176 UnsignedInt Protocol (R/O) = SIP String QCIF MPI (R/W) = 1 FMTP name: QCIF (33) UnsignedInt Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String Target Bit Rate (R/W) = 621700 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt
0:03.695 sipcmd OpalPlugin to_customised_options: Format Name = theora Media Type = video Payload Type = [pt=124] Encoding Name = theora CAP Delivery (R/W) = in_band FMTP name: delivery-method (in_band) String CAP Height (R/W) = 576 FMTP name: height (15) UnsignedInt CAP Sampling (R/W) = YCbCr-4:2:0 FMTP name: sampling (YCbCr-4:2:0) String CAP Width (R/W) = 704 FMTP name: width (15) UnsignedInt Clock Rate (R/O) = 90000 UnsignedInt Content Role (R/W) = No Role Enum Content Role Mask (R/W) = 0 UnsignedInt Frame Height (R/W) = 288 UnsignedInt Frame Time (R/W) = 1500 UnsignedInt Frame Width (R/W) = 352 UnsignedInt Max Bit Rate (R/W) = 768000 UnsignedInt Max Rx Frame Height (R/O) = 720 UnsignedInt Max Rx Frame Width (R/O) = 1280 UnsignedInt Max Tx Packet Size (R/O) = 1444 UnsignedInt Min Rx Frame Height (R/O) = 144 UnsignedInt Min Rx Frame Width (R/O) = 176 UnsignedInt Protocol (R/O) = SIP String Rate Control Enable (R/W) = 0 Boolean Rate Controller (R/W) = String Target Bit Rate (R/W) = 768000 UnsignedInt Tx Key Frame Period (R/W) = 125 UnsignedInt
0:03.696 sipcmd SIP PDU is too large (1747 bytes) trying compact form. 0:03.697 sipcmd SIP PDU is likely too large (1697 bytes) for UDP datagram. 0:03.697 sipcmd SIP Sending PDU (1697 bytes) to: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0 INVITE sip:503@192.168.4.203 SIP/2.0 CSeq: 1 INVITE v: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bKa2932bfa-752b-e311-8d69-f9c4e96053ec;rport User-Agent: sipcmd/1.0.1 f: "root" sip:599@192.168.4.203;tag=cee129fa-752b-e311-8d69-f9c4e96053ec i: 20ea29fa-752b-e311-8d69-f9c4e96053ec@ubuntu k: 100rel,replaces Organization: Command line VoIP testphone t: sip:503@192.168.4.203 m: "root" sip:599@192.168.4.208 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK l: 1141 c: application/sdp Max-Forwards: 70
v=0 o=- 1380899945 1 IN IP4 192.168.4.208 s=sipcmd/1.0.1 c=IN IP4 192.168.4.208 t=0 0 m=audio 5000 RTP/AVP 123 125 3 116 117 118 119 0 8 9 121 122 103 115 104 102 114 101 100 a=sendrecv a=rtpmap:123 AMR-WB/16000/1 a=fmtp:123 octet-align=1 a=rtpmap:125 AMR/8000/1 a=rtpmap:3 gsm/8000/1 a=rtpmap:116 G726-40/8000/1 a=rtpmap:117 G726-32/8000/1 a=rtpmap:118 G726-24/8000/1 a=rtpmap:119 G726-16/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:9 G722/8000/1 a=rtpmap:121 G7221/16000/1 a=fmtp:121 bitrate=24000 a=rtpmap:122 G7221/16000/1 a=fmtp:122 bitrate=32000 a=rtpmap:103 SILK/16000/1 a=rtpmap:115 Speex/16000/1 a=rtpmap:104 lpc10/8000/1 a=rtpmap:102 SILK/8000/1 a=rtpmap:114 Speex/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=maxptime:22 m=video 5002 RTP/AVP 31 97 124 b=AS:186624 b=TIAS:186624000 a=sendrecv a=rtpmap:31 h261/90000 a=fmtp:31 CIF=1;QCIF=1 a=rtpmap:97 raw/90000 a=fmtp:97 rate=90000;height=288;width=352;colorimetry=BT601-5;depth=8;sampling=YCbCr-4:2:0 a=rtpmap:124 theora/90000 a=fmtp:124 height=576;width=704
0:03.697 sipcmd OpalUDP Setting interface to 192.168.4.208%venet0:0 0:03.697 sipcmd SIP Transaction timers set: retry=10.000, completion=32.000 0:03.697 sipcmd OpalCon SetPhase from UninitialisedPhase to SetUpPhase for Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalCon OnSetUpConnectionCall[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalEP OnSetUpConnection Call[C49c691cd1]-EP[cee129fa-752b-e311-8d69-f9c4e96053ec]
0:03.697 sipcmd OpalMan SetUpCall succeeded, call=Call[C49c691cd1]
0:03.924 Opal Listener:0x8a73700 OpalUDP Binding to interface: 192.168.4.208:5060
0:03.924 Opal Listener:0x8a73700 SIP Waiting for PDU on udp$192.168.4.203:5060
0:03.925 Opal Listener:0x8a73700 SIP PDU received: rem=udp$192.168.4.203:5060,local=udp$192.168.4.208:5060,if=192.168.4.208%venet0:0
SIP/2.0 404 Not found
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 192.168.4.208:5060;branch=z9hG4bK8a9429fa-752b-e311-8d69-f9c4e96053ec;received=192.168.4.208;rport=5060
User-Agent: Asterisk PBX
From: sip:599@192.168.4.203;tag=524029fa-752b-e311-8d69-f9c4e96053ec
Call-ID: a22c29fa-752b-e311-8d69-f9c4e96053ec@ubuntu
Supported: replaces
To: sip:599@192.168.4.203;tag=as432f22a7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Content-Length: 0