tag:blogger.com,1999:blog-39820060226133240092024-03-08T08:59:16.835-08:00Practicing Voice over IP TechnologyVenkyhttp://www.blogger.com/profile/00292303755961414796noreply@blogger.comBlogger4125tag:blogger.com,1999:blog-3982006022613324009.post-80404516566910859362011-04-05T20:15:00.000-07:002015-08-19T20:55:27.880-07:00CUCM Notes.<div dir="ltr" style="text-align: left;" trbidi="on">
<strong><u><em><span style="color: yellow;">CUCM groups Advantages:</span></em></u></strong><br />
1. Redundancy<br />
2. Load Balancing. <br />
<br />
During failover from one call manager to another call manager in the CUCM group list, all active calls would be preserved, and Cisco IP phones would re-register when the existing calls are complete.<br />
<br />
Only on Call Manager group can be used for autoregistration per cluster.<br />
<br />
<strong><em><u><span style="color: yellow;">Connection Monitor Duration:</span></u></em></strong><span style="color: blue;"> </span><br />
The length of time before a phone that is using SRST fails back to a Call Manager after it has become available again.<br />
<br />
<strong><u><em><span style="color: yellow;">Device Pool:</span></em></u></strong><br />
Easy way of setting up same parameters for a set of devices (phones, gateways etc.,)<br />
<br />
<strong><u><em><span style="color: yellow;">Calling Search Space:</span></em></u></strong><br />
If you configure the calling search space on both device & line, then Call Manager concatenates both calling search spaces & puts line calling search space first.<br />
<br />
<strong><u><em><span style="color: yellow;">Media Resource Group List (MRGL):</span></em></u></strong><br />
If a device has MRGL configured at device level as well as at device pool level, then when a Media Resource is required Call Manager first searches the MRGL of device then MRGL at device pool.<br />
<div>
</div>
<ul>
<li>If a Media resource (transcoder, MTP or Conference bridge) is not assgined to any MRG then it belongs to default MRG & can be accessed accessed by Call Manager as a last resort. This media resource can be used by any device in the Call Manager.</li>
<ul>
<li>device MRGL ---> device pool MRGL ----> default MRGL.</li>
</ul>
</ul>
</div>
Venkyhttp://www.blogger.com/profile/00292303755961414796noreply@blogger.com0tag:blogger.com,1999:blog-3982006022613324009.post-8347424569312788242011-03-19T22:58:00.000-07:002015-08-19T20:56:14.066-07:00MGCP Vs H.323<div dir="ltr" style="text-align: left;" trbidi="on">
<strong><u><em><span style="color: yellow;">MGCP</span></em></u></strong> - IETF standard = Media Gateway Control Protocol. <br />
<ul style="text-align: left;">
<li> Master/Slave Protocol - Call Agent (CUCM) is used to control the gateway ports.</li>
<li>Complete control of dial-plan from the CUCM.</li>
<li>Centralized call processing (CUCM). * Requires dial-peers only for SRST.</li>
<li>Survivable end point by default- PRI backhaul.</li>
<li>No Caller-ID support with FXO port.</li>
</ul>
<br />
<span style="color: yellow;"><strong><u><em>H.323</em></u></strong> </span>- ITU standard = ITU Umbrella recommendation.<br />
<ul style="text-align: left;">
<li>Peer to Peer Protocol.</li>
<li>dial-plan controlled locally from H.323 gateway</li>
<li>distributed call processing. CUCM/PSTN sends call to gateway & gateway decides (based on dial-peer) which port/call manager to use for outbound/inbound calls. </li>
<li>Caller -ID supported with FXO port.</li>
<li>Not a survivable end point by default. Survivability (call preserving) is supported form 12.4(9T) IOS release. Requires manual configuration, as listed below. Also requires CUCM service parameter to set for "<span style="color: yellow;">Allow peer to Preserve H.323 calls</span>"</li>
<ul>
<li><span style="color: yellow;">#voice service voip</span></li>
<li><span style="color: yellow;">#h323</span></li>
<li><span style="color: yellow;">#call preserve </span></li>
</ul>
</ul>
<br />
<br />
<strong><u><em>Survivability (Call Preserving):</em></u></strong><br />
<br />
When there is WAN connection failure or degraded WAN connection between gateway & CUCM or ccm.exe service is stopped on the call manager - <br />
<br />
if the gateway is MGCP, all active calls would be preserved, but all new calls tried would be failed & supplementary services like hold, transfer would not function until gateway registers with secondary Call Manager. The MGCP gateway would try to re-register with secondary Call Manager in the CUCM group. When the gateway re-registers with secondary Call Manager, call preservation involves three steps which are completey transperent to the users.<br />
<ol>
<li>CUCM sends AUEP to the gateway - to find status of each B-channel on gateway.</li>
<li>CUCM sends AUCX to end point for which gateway reportes as a preserved call.</li>
<li>Finally Q.931 status enquiry message to confirm the status of calls that CUCM believes are preserved.</li>
</ol>
if the gateway is H.323, all active calls would be lost (if call preserve is not configured). Because as soon as CUCM detects TCP connection lost it cleares all calls & closes the TCP sockets for all active calls by sending TCP FIN message. If the gateway came up within a short while the gateway would receive the TCP FIN message and clears all calls. If the gateway took a while to come up, the TCP FIN message will not reach the gateway. But gateway would try sending keepalive messages to the CUCM for restoring the connection with primary Call Manager.When the Call Manager receives these keepalive messages it sends TCP RST (reset - tear down all active calls) message in response as it closed all TCP sockets. So either way H.323 gateway could not preserve any active calls during failover & fail back scenarios.<br />
<br />
If the "<span style="color: yellow;">call preserve</span>" is configured (supported on IOS versions starting 12.4(9T)) this command causes H.323 gateways to ignore socket closure messages (socket error) on H.225 & H.245 connections for active calls. That is how H.323 gateway preserves active calls using call preserver configuraiton.<br />
<br />
<strong><u><em><span style="color: yellow;">PRI backhaul.</span></em></u></strong><br />
Transporting of signalling information from MGCP gateway to Call Control Agent (CUCM) is called PRI backhauling.<br />
<ul>
<li>PRI backhauling is carried over TCP port 2428.</li>
<li>All Q.931 messages & D-channell signalling information is passed between CUCM & gateway using this TCP connection.</li>
<li>All L2 information is terminated at the gateway & L3 information is passed to CUCM - Call Agent.</li>
<li>MGCP gateway does not parse or have any knowledge of these signalling packets.</li>
</ul>
<br />
<strong><u><em>Useful Q.931 timers</em></u></strong><br />
<br />
T310 - how long to wait to get response such as ALERTING or CONNECT<br />
T303 - how long to wait to get response such as Call Proceeding for SETUP message.<br />
<br />
<br /></div>
Venkyhttp://www.blogger.com/profile/00292303755961414796noreply@blogger.com2tag:blogger.com,1999:blog-3982006022613324009.post-4661430734381187902011-01-25T20:45:00.000-08:002015-08-19T20:58:22.851-07:00Xcoder & Conf Br<div dir="ltr" style="text-align: left;" trbidi="on">
<strong><u><em>Configuring Conference Bridge & Transcoder - Gateway (CLI) part</em></u></strong><br />
<br />
<strong><u><em>Definitions/Usage : </em></u></strong><span style="color: blue;">"</span><span style="color: yellow;">dspfarm" vs "dsp services dspfarm</span><span style="color: blue;">"</span> <br />
<span style="color: yellow;">dspfarm</span><span style="color: blue;">:</span> Allows dsp resources pooling or sharing. That is, a VWIC card in a NM without any dsp resources can use mother board dsp resources.<br />
<br />
<span style="color: yellow;">dsp services dspfarm:</span> Allows any unused dsp resources to be allocated for conferencing or transcoding. Atleast one voice-card should be configured/enabled for this service.<br />
<br />
<br />
<strong><u><em>Configuration:</em></u></strong><br />
<br />
<span style="color: yellow;">sccp local looback X or gig X/X</span> ---> specifies the local interface that SCCP applications use to register with Call Manager.<br />
<br />
<span style="color: yellow;">sccp ccm x.x.x.x identifier y version X.X</span>---> adds the specified call manager to the list of available servers. Identifier identifies the call manager with a number in the list. Required once for each call manager server in the list. For better failover stategy keep the servers in the same order of the device pool (with which this application registers in CUCM). Version is the call manager version to which it registers. <br />
<br />
<span style="color: yellow;">sccp ip precedence X (1 - 7)</span> ---> sets the predence value to be used by SCCP application. <br />
Range 1 -7 & default value is 5.<br />
<br />
<span style="color: yellow;">sccp</span> ---> enables SCCP protocol and its related applications (transcoding/conferencing).<br />
<br />
<br />
<strong><u><em>dspfarm profile:</em></u></strong><br />
<br />
<span style="color: yellow;">dspfarm profile XX conference/transcode</span> ---> enters the dspfarm profile mode. Identifies the profile number (XX)& if it is for conferencing or for transcoding.<br />
<br />
<span style="color: yellow;">codec XXXXXX -</span>--> specify all the codecs you want this application to support. Ex: g711ulaw/g729abr8<br />
<br />
<span style="color: yellow;">maximum sessions XX </span>---> specifies the maximum number of sessions supported by this profile. It depends on the available registered dsp resources. Gateway automatically calculates and shows you the minimum & maximum number sessions (with maximum sessions ?) it can support depending upon the codecs you selected in above step.<br />
<br />
<span style="color: yellow;">associate application SCCP </span>---> Associates the SCCP protocl to the dspfarm profile (Application).<br />
<br />
<span style="color: yellow;">no shutdown</span> ---> enables/activates this application/ dspfarm profile.<br />
<br />
<br />
<strong><em><u>ccm group:</u></em></strong><br />
<br />
<span style="color: yellow;">sccp ccm group XXX</span> ---> enters the SCCP call manager config mode and creates call manager group<br />
<br />
<span style="color: yellow;">associate ccm X priority Y</span> ---> adds specified call manager to the group and defines its priority in group.<br />
Repeat for each call manager server defined in the call manager list (above).<br />
<br />
<span style="color: yellow;">associate profile XX register Name</span> ---> associates above defined dspfarm profile with call manager group and registers with the call manager with specified name (Name must match with the device name specified in the call manager otherwise it will not register.)<br />
<br />
<span style="color: yellow;">bind interface loopback X or gig X/X</span> ---> binds an interface with call manager group. <br />
This interface is used for signalling/media whereas "sccp local interface" command used for registration.<br />
<br />
<strong><u><em>Example Config:</em></u></strong><br />
<br />
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp local Loopback0</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp ccm 10.10.1.1 identifier 1 priority 1 version 7.0 </div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp ccm 10.10.2.2 identifier 2 priority 2 version 7.0 </div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp ccm 10.10.2.3 identifier 3 priority 3 version 7.0 </div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp ip precedence 1</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
!</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
sccp ccm group 118</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>description Mexico Conference Bridge </div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>bind interface Loopback0</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>associate ccm 1 priority 1</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>associate ccm 2 priority 2</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>associate ccm 3 priority 3</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
associate profile 21 register ABCXCODE</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
!</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
dspfarm profile 21 transcode<span style="mso-spacerun: yes;"> </span></div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g711ulaw</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g711alaw</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g729ar8</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g729abr8</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g729r8</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>codec g729br8</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>maximum sessions 10</div>
<div style="font-family: "Times New Roman"; font-size: 11pt; margin: 0in;">
<span style="mso-spacerun: yes;"> </span>associate application SCCP</div>
<br />
<br />
<br />
<br />
<br />
<br />
<br />
<br /></div>
Venkyhttp://www.blogger.com/profile/00292303755961414796noreply@blogger.com0tag:blogger.com,1999:blog-3982006022613324009.post-77402376995678557302011-01-17T20:58:00.000-08:002015-08-19T20:59:04.369-07:00csim start<div dir="ltr" style="text-align: left;" trbidi="on">
<strong><u><em>Command #1 - csim start "dial string"</em></u></strong><br />
<br />
csim start "dial string" simulates a call to the specified dial string. Most useful command in testing dial-plans on Voice Gateways & CMEs.<br />
<br />
<strong><u><em>Need to remember when using csim start:</em></u></strong><br />
1. It works only with telnet & does not work with SSH1 or SSH2. You would see following error message when you try this command with SSH1 or SSH2 login: <br />
<br />
Router#csim start 911234567890<br />
csim: called number = 911234567890, loop count = 1 ping count = 0<br />
<span style="color: yellow;">csim err:csim_do_test invalid major major(16) minor(0)</span><span style="color: blue;"><br />
</span>csim: loop = 1, failed = 0 <br />
csim: call attempted = 1, setup failed = 0, tone failed = 0<br />
2. It works only on the gateway with dial-plan configured.<br />
<br />
3. When the destination phone (specified dial string) rings you can answer the phone but you would not hear anything.<br />
<br />
4. Can be used in conjunction with debug commands (debug q931 or debug ccapi etc.,)<br />
<br />
<br />
<strong><u><em>Examples:</em></u></strong><br />
<br />
<span style="color: yellow;">1. Successful call</span><br />
Router#csim start 919724156038<br />
csim: called number = 919724156038, loop count = 1 ping count = 0<br />
csim: loop = 1, failed = 0 <br />
csim: call attempted = 1, setup failed = 0, tone failed = 1<br />
<br />
2. <span style="color: yellow;">Unsuccessful call </span>(call placed to unknown phone # or extension that doesnot exist)<br />
Router#csim start 923456781234<br />
csim: called number = 923456781234, loop count = 1 ping count = 0<br />
<span style="color: yellow;">csim err csimDisconnected recvd DISC cid(20237) </span><span style="color: blue;"><br />
</span>csim: loop = 1, failed = 1 <br />
csim: call attempted = 1, setup failed = 1, tone failed = 0<br />
3. Sometimes even the call (test) is successful still setup failed shows 1 - And I really dont know the reason :(<br />
Router#<span style="color: yellow;">csim start 919724156038csim: called number = 919724156038, loop count = 1 ping count = 0</span><br />
*Jan 17 2011 11:55:42.785 MST: ISDN Se0/1/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 3<br />
*Jan 17 2011 11:55:42.785 MST: ISDN Se0/1/0:23 Q931: Sending SETUP callref = 0x0C9A callID = 0x8CFA switch = primary-5ess interface = User <br />
*Jan 17 2011 11:55:42.785 MST: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0C9A <br />
Bearer Capability i = 0x8090A2 <br />
Standard = CCITT <br />
Transfer Capability = Speech <br />
Transfer Mode = Circuit <br />
Transfer Rate = 64 kbit/s <br />
Channel ID i = 0xA98397 <br />
Exclusive, Channel 23 <br />
Called Party Number i = 0xA1, '19724156038' <br />
Plan:ISDN, Type:National<br />
*Jan 17 2011 11:55:42.837 MST: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8C9A <br />
Channel ID i = 0xA98397 <br />
Exclusive, Channel 23<br />
*Jan 17 2011 11:55:43.457 MST: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x8C9A <br />
Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have in-band info <br />
*Jan 17 2011 11:55:48.753 MST: ISDN Se0/1/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x8C9A<br />
*Jan 17 2011 11:55:55.377 MST: ISDN Se0/1/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x8C9A<br />
*Jan 17 2011 11:55:55.381 MST: %ISDN-6-CONNECT: Interface Serial0/1/0:22 is now connected to 19724156038 N/A<br />
*Jan 17 2011 11:55:55.381 MST: ISDN Se0/1/0:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x0C9A<br />
<span style="color: yellow;">csim err csimDisconnected recvd DISC cid(17292) <br />
csim: loop = 1, failed = 1 <br />
csim: call attempted = 1, <strong><u><em>setup failed = 1</em></u></strong>, tone failed = 0</span></div>
Venkyhttp://www.blogger.com/profile/00292303755961414796noreply@blogger.com0