RTP Media Stream with "Third Party SIP Device" always through CUCM
Hello,
i have i quite strage problem on one of my customers locations:
we have a cucm 7.1.5(SU4) with the cucm in the datacenter. And we have a small location(branch office) which has a small wan connection to the datacenter (1MBit/s).
In this location we have several Kirk (Polycom) Dect phones which register as "Third Party SIP Device - Basic" on the CUCM.
(The problem is the same if i use the X-Lite SIP Client instead)
When this SIP Phones or the X-Lite Client dials a internal Number of the same location the RTP Media Streams goes directly from the SIP Client to the phone. But if they dial an external number the RTP Stream goes from the SIP Client via the wan connection to the CUCM and back via the wan connection to the 2901 H.323 Gateway (on the same location).
and of course if i now start a big download or upload i'm no longe able to complete the phone call because we have no QOS on the wan connection, because we don't want to make calls over this connection.
When i look at the Sniffer files with Wireshark i see that the CUCM sends his own ip adresse in the SDP Header for the RTP Stream to the SIP Client. And this of course is wrong because the RTP Stream should always reside on the branch office.
i tested this in my lab (CUCM 8.5) and it is the same. i used the "Standard SIP Profile" and a Basic Third Party SIP Device"
The SCCP Phones on the same location which are configured with the same Region, Location, Device Pools, Media Resources and which use the same Gateway for external calls do not have this problem.
In the Gateway configuration "MTP Required" is not activated and i tested it in my lab with some Cisco SIP Phones (9971) and they are also not affacted with this problem.
any ideas?
do you have a SIP trunk to the external devices with MTP required checked ?
Similar Messages
-
CUOM with Third-party SIP Device
Hi everyone,
I have installed CUOM in environment with CUCM, Cisco IPPhones (SCCP), and IPPhones (Third-party SIP Device), I can monitor CUCM and SCCP IPPhones but the Third-party SIP Device can't be monitored with CUOM, Please someone have a solution for this issue
Thanks in advance,do you have a SIP trunk to the external devices with MTP required checked ?
-
Activate a call forward with a Third-party SIP Device or with a analog device
Hi,
In a CUCMv9, how i can activate a call forward (all, busy, no anwser...) with Third-party SIP Device or with a analog device connected to a fxs?
I want to activate a call forward like a Alcatel or Aastra PBX with a code.
For exemple, i pick up the phone, with the code *95 followed by the destination number and hangs up the phone. And use the #95 for désactivate this call forward.
It's possible?
Thanks.No codes for 3rd party SIP phones, no way to do it. Or for that matter, not even for Cisco Phones, other than CFA.
Anything besides CFA needs to be done via CCMadmin or CCMuser for any kind of phone.
For FXS that's only doable if you're running SCCP
http://www.cisco.com/en/US/partner/docs/ios/voice/fxs/configuration/guide/fxssccpsplmft.html
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk -
Third-party SIP Device (Basic) video calling
Hi,
I want video calling with third-part SIP device to register CUCM 8.6. But my sip phones rejecting video calls. (there is no problem, audio calling...)
I'm doing the same process with elastix ip pbx smoothly.
Help me !!ip phone codec h264 , h263
-
How to communicate with third party bluetooth device using LAbVIEW
Hi
I am trying to communicate with a third party bluetooth device using LabVIEW, I am using a bluetooth dongle. I am able to discover the device and able to open the connection by specifying the service as serial port. But after that when I am trying to read and write data between LabVIEW and the bluetooth device (using Bluetooth Read & Bluetooth Write functions) its not happening.
Dose anybody faced same kind of issue if so kindly guide me, this is first time I am using Blutooth protocol.
Regards
VisumanHi,
I am able to talk with bluetooth / serial devices by using Labview protocol (not Wii dll). These are steps:
Discover the bluetooth device in range (Bluetooth dicover.vi)
Get the address of the device
Pass device address to Bluetooth RFCOMM Service Discovery.vi to get service list
Pass channel and uuid to Bluetooth open connection
After that do write and read (Bluetooth Write / Bluetooth read).
I hope it works for you.
Take care
tp
-
Paging Third Party SIP Phones connected to CUCM
Current SetUp: CUCM - Cisco 3925, Two MCS 7816 (Call Control Server) and One MCS 7825 (Voice mail server)
We have third party SIP phones configured in auto answer mode. These phones are used to make live announcements.
To Do:
There are approximately 80 phones in the system and the requirement is to select any combination of these phones to make Public announcement (or Paging).
Is there an application that enables us to select any combination of phones on the fly to do paging? How can we select a mp3 file to play on a phone in an auto answer mode?
Any help will be appreciated.
Thanks
SidThere's nothing built into call manager to do this. You could investigate using the Cisco Unified Application Environment (CUAE) and write a script to do this, or there are some 3rd party applications that might work for you such as Berbee's Informacast.
-
Dear All,
We have two cucm Clusters in Different Locations between that clusters i created
Inter-Cluster Trunk (Non-Gatekeeper Controlled) Now all are working fine Between Clusters
audio calls & Video calls between sccp 8945 phones , but iam facing a Problem with third party
Video Phones (Polycom VVX 1500 ) Third Party SIP Phones located in second cluster, From 1 st cluster cisco 8945 Video
phone to 2nd cluster Polycom Video phone all calls are works for voice call only, but no video ,
Please Suggest me Solution.
Thank you,
SrimanTry setting up a SIP trunk between the two clusters and set a route patten just to the VVX 1500 and check how that goes.
From memory inter-cluster trunks are a H.323 like protocol which might have video inter-op issues with the Polycom device. -
Add third party SIP Phone to CCM 5
'm not able to register this SIP Phone to the CCM5.0. I have device license that cater all IP Phone models.(LIC-CM-DL-100=)
I got error message " Login Forbidden" "timeout" in the IP Phone.
In the CCM, I got this message in Phone COnfig Window
Registration: Rejected.
Can you explain on how to register this 3rd party IP phone to CCM?
Is it CCM able to support SIP Phone?Hi,
This is most likely because of the following...
Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
The REGISTER message includes the following header:
Authorization: Digest username="swhite",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
The username, swhite, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
See the following document.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
Hope this helps, if so please rate.
Regards,
Dave -
MOH for third party sip phones
Hello ,
I using CUCM version 9.1.1.2000-5 .
Does this version support MOH for third party sip phones ?
Thank youHi,
I couldn't found better piece of information which list all cases how MoH is implemented for various terminals and hence thought of testing the same. My observations;
SCCP phone -> Hold -> SIP Phone -> MoH plays to SIP phone
SIP Phone -> Hold -> SCCP phone - > Doesn't play MoH
SIP Phone -> Hold -> SIP phone -> Doesn't play MoH
Please note that I have checked with both Xlite and 3CX, results are same.
I have verified in wireshark also, call manager is not sending RTP packets to held party when call is hold by third party SIP phone.
Checked in CUCM 9.1
Thanks
Vivek -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
Conflict with Third Party Apps?
I am getting an occasional error saying there is a possible conflict with third party apps and that I need to check the drivers of my MIDI devices to see if they are up to date? Sorry but I did not record the exact wording. Does anyone know what this might mean, and how I need to go about checking drivers? I have a lot of interfaces and not sure how to see what's what. Thanks.
Hi Midlake,
Well, because they are known troublemakers. Search for 'takes' or 'take folder(s)' or 'comp' here on the forum and you'll find many a troubled soul crying out for help - including seasoned pro's, who thought they were beyond tears...
O, they are fine for recording many takes - but thats' all. I'll select the best takes/phrases.
regards, Erik. -
Hi to all,
Can any one share the configuration of how to connect the SAP HR with the Third party time device of swipe in and swipe out of an employee.
You can mail the document to my id of shreyasen1980 at the rate of gmail dot com
Thanks,
ShreyasenHi,
There are many third party devices that can be integrated with SAP for capturing time events in IT2011. Timelink is one of them and it is SAP certified partner. Following are the high level steps to be followed considering SAP integration with Timelink.
Setup and assign timelink user with proper profiles
Setup RFC destination for the timelink
Create the logical system for timelink
Define ports for IDOC processing
Define the distribution model for timelink for exchanging inbound and outbound messages
Define the partner profiles
Verify the IDOC list and the communication manually between Timelink and SAP
Best Regards,
Sunny -
CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls
Hi Team,
we are running CUCM 9.1(2a),
we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
Regards
AnanthakumarAre A and B both Avaya phones?
So it looks like you're not seeing the alerting name/connected name getting updated then? Do you have alerting names configured on the directory numbers? Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it. Might just be something that needs to be tweaked in the 46xxsettings.txt file. -
MainStage message: possible conflict with third party midi or audio drivers
Just when I thought everything was going to be ok...
MainStage gives me a message that it detects a possible conflict with third party audio or midi drivers. The only midi device I'm using is class compliant (doesn't need a driver). My audio interface is an Apogee Duet and I'm using the most recent driver. Logic registers no such complaint. MainStage freezes. Any ideas?
Thanks,
MarkHello.
I am seeing this whenever MIDI events are being sent into either Logic or MainStage (primarily MainStage), while they are launching. On my setup, this will either crash the application during launch, ( sent to Apple ) or trigger this "...3rd party conflict / driver..." dialog.
In either instance, disabling incoming MIDI messages by turning them off on all connected MIDI devices, or as suggested, waiting to connect the device in question until, the application is fully launched, fixes my version of this issue.
** Please Note: In my experience, increasing the number incoming MIDI messages, exacerbates the issue and, makes a dialog or crash more likely to occur.
Hope this is helpful. -
Injecting DTMF event in the midst of RTP media streaming?
Hi all, I am attempting to implement a mechanism that allows me to inject DTMF RTP events in an RTP media stream. This is useful in telephony applications where the users are prompted to enter digits while being served by automated voice services such as answerinng services or tele-bankings etc. So basically my approach is to extend Sun's RTP packetizer and intercept outgoing packets when necessary and replace them with the appropriate DTMF RTP (RFC 2833) packets. Sounds simple enough. So I derived my custom packetizer from com.sun.media.codec.audio.ulaw.Packetizer, and over-ridden method process(). Normally my packetizer's method process() simply delegates the functionality to its parent class's process() method. When a DTMF digit is required I'd take the outputbuffer generated by the parent's process() and modify its header and payload to turn it into a DTMF RTP packet. The problem is that the class RTPHeader is so limited, there is no way to set the payload type, sequence number, ... And the documentation is precious few and far in between. If someone has solved this issue, or if you have a reference to some documentation that describes the inner working of JMF's codec chaining, I would appreciate some pointers. What I need to know is:
- What JMF does with the Buffer objects between stages (from one codec to the next)?
- The data portion of Buffer is an Object of arbitray class, what on earth does JMF do with that?
- How does JMF take a Buffer object and turn it into a UDP packet?
- How do I go about creating an RTP header and fill it with the information (payload type, timestamp, sequence number, ...) that I want?
Thanks in advance,I don't know what the heck RTPHeader represents but it sure doesn't seem to conform to RFC1889. And also, it seems the UDP RTP packets are formed somewhere after the packetizer and before the RTP connector, someone oughta jot all this down in a book. So anyway, using my own RTPConnector implementation I have some control over the outgoing/incoming RTP packets, to access and manipulate the real RTP header (not RTPHeader), I devised a new class that takes the RTP packet buffer and provides an API to examine and manipulate some RTP info directly on the buffer without doing any unecessary data copy (you guys can extend it to do more as per your requirement):
package com.mycompany.media;
import javax.media.rtp.RTPConnector;
import java.net.DatagramSocket;
import java.net.InetAddress;
import java.io.IOException;
import javax.media.protocol.ContentDescriptor;
import javax.media.protocol.SourceTransferHandler;
import java.net.DatagramPacket;
import javax.media.protocol.PushSourceStream;
import javax.media.rtp.OutputDataStream;
import java.net.SocketException;
import mitel.utilities.MiQueue;
import java.nio.ByteBuffer;
import java.util.LinkedList;
class MiRtpHeader
byte[] data;
int myoffset;
public MiRtpHeader(byte[] buf, int offset, int len) throws ArrayIndexOutOfBoundsException
if(len < 12)
throw new ArrayIndexOutOfBoundsException("Buffer not large enough to contain a basic RTP header");
data = buf;
myoffset = offset;
public boolean getExtension()
return (0 != (data[myoffset] & 0x10));
public void setExtension(boolean state)
if(state)
data[myoffset] = (byte)(data[myoffset] | 0x10);
else
data[myoffset] = (byte)(data[myoffset] & 0xef);
public boolean getMarker()
return (0 != (data[myoffset + 1] & 0x80));
public void setMarker(boolean state)
if(state)
data[myoffset + 1] = (byte)(data[myoffset + 1] | 0x80);
else
data[myoffset + 1] = (byte)(data[myoffset + 1] & 0x7f);
public int getTs()
ByteBuffer tsBuf = ByteBuffer.wrap(data, myoffset + 4, 4);
return tsBuf.getInt();
public void setTs(int ts)
ByteBuffer tsBuf = ByteBuffer.wrap(data, myoffset + 4, 4);
tsBuf.putInt(ts);
public int getPayloadType()
ByteBuffer ptBuf = ByteBuffer.wrap(data, myoffset + 1, 1);
return ptBuf.get();
}
Maybe you are looking for
-
How can I make a window transparent but keep its objects opaque in Cocoa-Applescript?
How can I do this in Cocoa-Applescript, Xcode? As well, is it possible to blur the area behind where the transparent window is?
-
As soon as I updated firefox, as prompted, it takes FOREVER to open a site, including when i first open firefox. Should I uninstall and then reinstall?
-
What's wrong with my iPhone 5c?
Last night I was on my phone until it died. It was working perfectly fine as usual. Then this morning when I went to charge it, it turned on, but the screen looked distorted and the touch was not working. The only thing I can do is turn it on and off
-
How do I add "Shared" folder to my Itunes Library
file://localhost/Users/jodyaltschule/Desktop/Screen%20shot%202011-08-24%20at%201 2.21.29%20PM.png.zip
-
Software Restriction Policy batch vs vbs
Hi there, I have recently implemented a Software Restriction Policy on a Computer level with Disallowed level as default. I whitelisted the \\mydomain\SysVol so that my Group Policies could run. I have a few batch files that run upon user logon. The