Problem 1: SIP trunks calls getting drops in Call Manager which is from Nortel to Cisco IP phones

If you are calling from third party IP phone to cisco IP phone or Vice versa. Call flow will be through SIP
Trunk. So call should flow through to the SIP trunk and for this purpose we need MTP (which is helpful
to direct the call to SIP Trunk) and we need Transcoder in this process bcoz there are two different IP
network region available. So, they will use different Codec from one region to another region and
transcoder will plays Important Role for changing two different codec. Then MRLG is Media group which
having MTP to be used for diff. purpose.

Main Reason for this problem is Codec Mismatch r Transcoder.

We’ll go by step by step process

S1: Check destination pattern of out bound call and dial peer conf. by using command Sh Voice port
summary

S2: Check MTP Required in SIP Trunk

S3: Check MTP Status using Sh Run , Sh MTP, Sh SCCP Connection in CME (Media Termination Point)

S4: Check DSP farm Profile using same above command in CME (Transcoder)

S5: Check Region in CUCM (Codec Problem)

S6: restart.

If there is no MTP in MRLG group then add MTP to the particular MRLG.

If all above things are correct mean then you need to check SIP profile in CME.

If SIP profile like this means you need add one more Command.

Voice class sip-profiles 1

request INVITE sip-header Allow-Header modify “, UPDATE” “”

voice class sip-profiles 1

request INVITE sip-header Allow-Header modify “UPDATE,” “” < < < This was already there
request REINVITE sip-header Allow-Header modify ” UPDATE,” “”
response 200 sip-header Allow-Header modify “UPDATE,” “”
response 180 sip-header Allow-Header modify “UPDATE,” “”

After that the next step was to increase the Service Parameter “SIP Session Expire Timer” which
you can find under Service Parameters > Call manager.
Creating Transcoders
Transcoders are hardware devices that convert calls from one codec to another. This
needs to be done when a call is placed between two devices that cannot communicate
using the same codec. For example, if a device that could only use G.729 called a device
that could only use G.711, a transcoder is necessary to convert the codec for the call to
take place. Here is a real-world example. A phone in a remote branch, which is configured
to use G.729 when placing calls across the WAN, wants to participate in a conference call
that originated on the other side of the WAN, and a software conference bridge is being
used. Because software conference bridges support only G.711 and the remote branch
0 Configuring Cisco Unified Communications Manager and Unity Connection: A Step-by-Step
Guide
can only support G.729 across the WAN, a transcoder is needed to convert G.729 to
G.711 and vice versa.
Transcoders can run on a number of Cisco devices. These devices have digital signal
processors (DSP) that can be used for transcoding purposes. The devices that support
hardware transcoding continue to expand. It is recommended that you check Cisco.com
for the most current list of hardware. The Cisco 2821 with an NM-HDV2 is an example
of the wide range of equipment that can serve as transcoders.
Depending on the hardware that is being used, the configuration of a transcoder will
vary. The main difference is the first field that you need configure. It will either be a
MAC address or the name of the device. To give you an idea of the configuration
process, the following steps illustrate how to configure a Cisco 2821XM with an
NM-HDV2 as a transcoder:
Step 1. From within CCMAdmin, select Media Resource > Transcoder.
Step 2. On the next page, click the Add New link.
Step 3. From the Transcoder Type drop-down list, select Cisco Conference
Enhanced IOS Media Termination Point.
Four different types of transcoders can be configured. Table 6-3 provides a
brief explanation of each with examples of the type of hardware required.
Step 4. In the Description field, enter a description that helps identify the purpose
of the conference bridge. In that description, you might want to include on
which Catalyst it is installed.
Table 6-3 Transcoders
Transcoder Type Feature Hardware
Cisco Media
Termination Point (WSSVC-
CMM)
Transcodes G.711, G.729, and G.723 WS-SVCCMMACT
Cisco Media
Termination Point
Hardware
Transcodes G.711, G.729, G.723, GSM FR, and GSM
EFR
WSX6608-
T1
WSX6608-
E1
Cisco IOS Media
Termination Point
Transcodes G.711 and G.729 NM-HDV
Cisco IOS Enhanced
Media Termination Point
Transcodes G.711 and G.729 NM-HD
NMHDV2
NM-HD-
1V/2V/2VE
Chapter 6: Configuring CUCM Features and Services 281
Step 5. Enter name of the IOS device in which the transcoding resource is installed in
the Device Name field.
Step 6. From the Device Pool drop-down list, select the device pool for the conference
bridge.
Step 7. From the Common Device Configuration drop-down list, select the common
device configuration that the device will use.
Step 8. The Special Load field enables you to enter special load information. In most
cases, this field is left blank.
Step 9. The Use Trusted Relay Point field determines whether a relay point such as a
Media Termination Point (MTP) or a transcoder must be labeled trusted to
be used by this device. This field is typically changed only in virtualized
environments.
Step 10. Click the Insert button to add the transcoder.
After the transcoder is created, it is available for use. Later in this chapter, you see how to
assign transcoders to media resource groups and lists.
 Problem 2: Adding E20 Cisco IP phone to call manager

TANBERG E20 is video conference IP phone (act like SIP phone in call manager 6&7) which can
be used three different SIP profile. So, we need to add this IP phone in call manager (6&7) as
third party SIP phone but Call manager 8 series add as Native IP Phone. Check there are two
procedure below.

Procedure for setting up a Third Party SIP device with CUCM (Mitel 5224 in this case):

CUCM one-time setup:

1. Create a SIP Profile for the third-party SIP phone which is a copy of the Standard SIP Profile
2. Create a copy of the Advanced SIP Phone Security Profile for the third-party SIP phone and check the Digest
Authentication box

CUCM setup per-phone:

1. Create the SIP Phone in CUCM as an Basic or Advanced SIP Device

( Basic requires 3 DLU's gives you 1 line. Advanced requires 6 DLU's and gives you up to 8 lines with video)
a. Use the SIP Profile and Security Profile you created earlier
b. Set the phone's digest user to the end-user of the phone
c. Create a line for the phone and note the extension
2. Find the end-user you want assigned to a phone in CUCM
a. Set their Digest Credentials (this can be bulk assigned to be the same password for all users but
is a security risk)
b. Associate the phone with the end-user normally

Third-party SIP phone setup per-phone (This is specific to a Mitel 5224 so do whatever is necessary for your
phone):

1. Boot phone while pressing the *superkey* (Blue button)

2. Use the buttons on the phone to set it into SIP mode and phone will reboot
3. Find Phone's IP address and login to phone admin page (u: admin p: 5224)
4. Select the Quick Start link on the left
a. User ID is the extension in Call Manager
b. User Display name is how you want the phone to display the number/user
c. SIP Authentication User and Password are the end-user's username and SIP Digest password in
CUCM
d. SIP Proxy server is the CUCM IP address
e. SIP Registry server is the CUCM IP address
f. Press Apply

From what I read above perhaps your security profile isn't setup correctly? Alternatively, if the Tandberg device
doesn't distinguish between the phone username and the authentication username then then the end-user's name in
CUCM must also be the extension number. If it does distingush between them you can use the phone username as
the extension (do not use the @<CUCMIPAddress>) and the authentication username as the end-user's username.
 Procedure for setting up a Third Party SIP device with CUCM (Mitel 5224 in this case):
CUCM one-time setup:
1. Create a SIP Profile for the third-party SIP phone which is a copy of the Standard SIP Profile
Dedevice-> device settings->sip profile

2. Create a copy of the Advanced SIP Phone Security Profile for the third-party SIP phone and check the Digest
Authentication box
SySystem->Security->Phone Security Profile tcp/udp, enable digest auth, port 5060

CUCM setup per-phone:

1. Create the SIP Phone in CUCM as an Basic or Advanced SIP Device (place device's MAC address in the Device
Name)
( Basic requires 3 DLU's gives you 1 line. Advanced requires 6 DLU's and gives you up to 8 lines with video)
a. Use the new SIP Profile and Security Profile you created earlier->protocol specific Information
b. Set the phone's digest user to the end-user of the phone->protocol specific Information
c. Create a line for the phone and note the extension
2. Find the end-user you want assigned to a phone in CUCM
a. Set their Digest Credentials (this can be bulk assigned to be the same password for all users but
is a security risk)
I set the user be the same as the extension
b. Associate the phone with the end-user normally

I configured the e20 phone with the extension@CUCM ipaddress format

Make sure to configure a proxy address and auth information in the settings proxy1,authentication etc of
the phone

Phone will register.

I am having some audio issues but at least the phone is registered

One more thing there are three different profiles are available.

For call manager 6 & 7 series E20 should use T.2.0 firmware

For call manager 8 Series E20 should use T.4.0 firmware.

So, it is not possible to add E20 phone in both call managers (Line 1 for CUCM 6&7 and Line 2
for CUCM 8) but we can add it in cisco call manager and third party call manager (like Nortel,
Alcatel etc..)(Line 1 for cisco call manager and Line 2 for third party call manager).
Understanding About DSP and PVDM

Voice termination applies to a call that has two call legs, one leg on a time-division multiplexing (TDM)
interface and the second leg on a Voice over IP (VoIP) connection. The TDM leg must be terminated by
hardware that performs coding/decoding and packetization of the stream. This termination function is
performed by **digital signal processor** (DSP) resources residing in the same hardware module, blade,
or platform. All DSP hardware on Cisco TDM gateways is capable of terminating voice streams, and
certain hardware is also capable of performing other media resource functions such as conferencing or
transcoding.

Conferencing

DSPs that are configured through Cisco IOS as conference resources will load firmware into the DSPs
that is specific to conferencing functionality only, and these DSPs cannot be used for any other media
feature.

The following guidelines and considerations apply to these DSP resources:

Based on the C5510 DSP chipset, the NM-HDV2 and the router chassis use the PVDM2 modules for
providing DSPs.

DSPs on PVDM2 hardware are configured individually as either voice termination, conferencing, media
termination, or transcoding, so that DSPs on a single PVDM may be used as different resource types.
Allocate DSPs to voice termination first, then to other functionality as needed.

The NM-HDV2 has 4 slots that will accept PVDM2 modules in any combination. The other network
modules have fixed numbers of DSPs.

A conference based on these DSPs allows a maximum of 8 participants. When a conference begins, all 8
positions are reserved at that time.

The PVDM2-8 is listed as having half a DSP because it has a DSP that has half the processing capacity of
the PVDM2-16. For example, if the DSP on a PVDM2-8 is configured for G.711, it can provide (0.5 * 8)
bridges/DSP = 4 conference bridges.

A DSP farm configuration in Cisco IOS specifies which codecs may be accepted for the farm. A DSP farm
that is configured for conferencing and G.711 provides 8 conferences. When configured to accept both
G.711 and G.729 calls, a single DSP provides 2 conferences because it is also reserving its resources for
performing transcoding of streams.

The I/O of an NM-HDV2 is limited to 400 streams, so ensure that the number of conference resources
allocated does not cause this limit to be exceeded. If G.711 conferences are configured, then no more
than 6 DSPs (total of 48 conferences with 8 participants each) should be allocated per NM because (48 *
8) participants = 384 streams. If you configure all conferencing for both G.711 and G.729 codecs, then
each DSP provides only 2 conferences of 8 participants each. In this case, it is possible to populate the
NM fully and configure it with 16 DSPs so there would be 256 streams.
Conferences cannot natively accept calls utilizing the GSM codec. A transcoder must be provided
separately for these calls to participate in a conference.

Any PVDM2-based hardware, such as the NM-HDV2, may be used simultaneously in a single chassis for
voice termination but may not be used simultaneously for other media resource functionality. The DSPs
based on PVDM-256K and PVDM2 have different DSP farm configurations, and only one may be
configured in a router at a time.

Transcoding

A transcoder is a device that converts an input stream from one codec into an output stream that uses a
different codec. It may also connect two streams that utilize the same codec but with a different
sampling rate. In a Unified CM system, the typical use of a transcoder is to convert between a G.711
voice stream and the low bit-rate compressed voice stream G729a. The following cases determine when
transcoder resources are needed:

Single codec for the entire system

When a single codec is configured for all calls in the system, then no transcoder resources are required.
The G.711 codec is supported by all vendors. A single-site deployment usually has no need for
conserving bandwidth, and a single codec can be used. In this scenario, G.711 is the most common
choice.

Multiple codecs in use in the system, and all endpoints are capable of all codec types

The most common reason for multiple codecs is to use G.711 for LAN calls to maximize the call quality
and to use a low-bandwidth codec to maximize bandwidth efficiency for calls that traverse a WAN with
limited bandwidth. Cisco recommends using G.729a as the low-bandwidth codec because it is supported
on all Cisco Unified IP Phone models as well as most other Cisco Unified Communications devices,
therefore it can eliminate the need for transcoding. Although Unified CM allows configuration of other
low-bandwidth codecs between regions, the current phone models do not support those codecs and
would require transcoders. They would require one transcoder for a call to a gateway and two
transcoders if the call is to another IP phone. The use of transcoders is avoided if all devices support and
are configured for both G.711 and G.729 because the devices will use the appropriate codec on a call-
by-call basis.
Multiple codecs in use in the system, and some endpoints support or are configured for G.711 only

This condition exists when G.729a is used in the system but there are devices that do not support this
codec, or a device with G.729a support may be configured to not use it. In this case, a transcoder is also
required. Devices from some third-party vendors may not support G.729. Another example where G.729
is supported but might not be configured is with Cisco Unity. Cisco Unity has support for accepting calls
with G.729a, but the codec is implemented in software and is CPU-intensive. Because as few as 10
simultaneous calls can cause significant CPU utilization, many deployments choose to disable G.729 on
Cisco Unity and off-load the transcoding function to a dedicated transcoding resource external to the
Unity server. If your system includes Cisco Unity, determine whether Unity will accept G.729a calls or
whether it will be configured for G.711 only.

To finalize the design, it is necessary to know how many transcoders are needed and where they will be
placed. If multiple codecs are needed, it is necessary to know how many endpoints do not support all
codecs, where those endpoints are located, what other groups will be accessing those resources, how
many maximum simultaneous calls these device must support, and where those resources are located in
the network

From the CUCM SRND;

This is a helpful item;

DSP Calculator

http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl