在IOS-XE语音网关上配置IP陆地移动无线电(LMR)/Hoot和Holler

简介

本文档介绍陆地移动无线电(LMR)或Hoot and Holler(Hootie)功能，该功能允许模拟设备通过启用组播的LAN与其他（模拟和IP）终端通信。

语音网关充当IP网络和模拟端点之间的分界点，促进模拟音频和组播实时传输协议(RTP)之间的会话。

作者：Kyzer Davis和Matt Snow，Cisco TAC工程师。

先决条件

要求

Cisco 建议您了解以下主题：

• 数字信号处理器(DSP)
• 模拟卡
• 功能的适用许可证

!
license boot level appxk9
license boot level uck9
! or
license boot suite FoundationSuiteK9
license boot suite AdvUCSuiteK9

• 支持组播的局域网或广域网

注意：本文档未涉及LAN或WAN上组播配置的许多方面。请参阅相关文档，以启用网络路径中LAN或WAN设备上的组播。

使用的组件

本文档中的信息都是基于特定实验室环境中的设备编写的。本文档中使用的所有设备最初均采用原始（默认）配置。如果您的网络处于活动状态，请确保您了解所有命令的潜在影响。

• 4451-X
• NIM-4E/M
• IOS-XE 16.3或更高版本。(发行说明)[推荐:IOS-XE 16.7或更高版本]

ISR4451# show inventory
NAME: "Chassis", DESCR: "Cisco ISR4451 Chassis"
PID: ISR4451-X/K9      , VID: V03  , SN: XXXXXXXXX

NAME: "NIM subslot 0/3", DESCR: "NIM-4E/M Voice Analog Module"
PID: NIM-4E/M          , VID: V01  , SN: XXXXXXXXX

注意：在ISR 4000语音网关上使用模拟NIM卡，使用本地DSP。 因此不需要主板DSP。

背景信息

潜在使用案例：

• 包括一键通设备的无线电设备和警报
• 组播信息通告（无线广播）
• 模拟炮塔系统

注意：以下是一些示例使用案例。应用程序不限于这些功能。

最初的LMR设计指南不包括最新一代思科语音网关所需的项目。因此，本文档旨在介绍有关IOS-XE设备（如ISR 4300和4400系列语音网关）的LMR/Hootie功能。

以下是示例拓扑；

第7层信令和媒体

Analog Endpoint <> Ear and Mouth (E&M) Port <> Cisco Voice Gateway (4451-X) <> Multicast Enabled LAN <> IP Endpoint.

提示：请记住，由于IP主干使用组播，语音网关只需能够成功加入所需的组播组。语音网关并不了解其他终端，也不直接与其通信，因此本文档详细说明了一个LMR/Hootie语音网关中的示例配置、调试、显示命令和故障排除。

配置

步骤1.您必须首先配置运行语音和组播功能所需的IOS-XE许可证。

config t
!
license boot level appxk9
license boot level uck9
! or
license boot suite FoundationSuiteK9
license boot suite AdvUCSuiteK9
!
exit
!
wr
!
reload
!

设备重新通电后，验证许可证状态是否与以下show命令输出匹配：

ISR4451# show license feature
Feature name             Enforcement  Evaluation  Subscription   Enabled  RightToUse
appxk9                   yes          yes         no             yes      yes
uck9                     yes          yes         no             yes      yes

步骤2.接下来，定义包含所需组播IP和端口的组播IP拨号对等体；

!
dial-peer voice 33333 voip
 destination-pattern 33333
 session protocol multicast
 session target ipv4:239.X.X.X:21000
 codec g711ulaw
 vad aggressive
!

Dial-peer命令语法：

步骤3.为便于组播组与模拟端口的此语音网关之间的永久（始终开启）连接，必须定义voice-class permanent，然后将其应用于语音端口。

!
voice class permanent 1
 signal timing oos timeout disabled
 signal keepalive disabled
!
voice-port 0/1/0
 voice-class permanent 1
!

voice class permanent命令语法

然后为E&M端口配置所需类型的连接语音端口，然后配置命令。(E&M或本文档未涵盖的其他模拟特定配置。

步骤4. Cisco hoot and holler over IP提供永远在线的通信网桥。最终用户无需拨打任何电话号码即可与热键组的其他成员联系。为了模拟此功能，Cisco IOS提供了称为Connection Trunk的功能。连接中继提供永久语音呼叫，不需要最终用户进行任何输入，因为所有数字均由路由器/网关内部拨号。

此连接中继将语音端口绑定到您在拨号对等体配置步骤中配置的组播地址。

!
voice-port 0/1/0
 connection trunk 33333
!

模拟端口命令语法

步骤5.语音配置完成后，您需要定义组播配置。

!
ip multicast-routing distributed
!
interface GigabitEthernet0/0/1
 ip address Y.Y.Y.Y 255.255.255.0
 ip pim sparse-mode
!
interface Vif1
 ip address 192.0.2.2 255.255.255.0
 ip pim sparse-mode
! 
interface Service-Engine0/1/0
 ip pim sparse-mode
!
ip pim rp-address 2.x.x.x
!

有关组播配置的说明：

• 服务引擎接口是模拟NIM上PVDM的第3层接口。这需要像任何其他入口/出口第3层接口一样，使用协议无关组播(PIM)命令进行配置
• 此 服务引擎不需要IP地址
• PIM配置的类型取决于LAN上的组播实施类型
• 必须启用组播路由，即使所有流量都在同一个VLAN中
• 对于源自路由器的组播RTP，IP必须是VIF IP减去1。因此我们的源必须为192.0.2.1，因为我们已在VIF上配置了192.0.2.2
  • 在某些情况下，这可能是VIF加1，但是对于此配置，VIF假设减1。请始终选中show ip mroute以查看路由器正在使用哪个VIF。
• 组播PIM RP可以是同一个语音网关，但是对于本实验，组播PIM RP位于通过EIGRP获知的网络(2.x.x.x)中的另一台设备上（未显示）

验证

使用本部分可确认配置能否正常运行。

语音验证

当配置完成时，永久连接将恢复正常。您可以使用此show命令输出进行验证；

ISR4451# show call active voice compact
 <callID>  A/O FAX T<sec> Codec       type        Peer Address       IP R<ip>:<udp>        VRF
Total call-legs: 2
       115 ANS     T24    g711ulaw    TELE        P
       116 ORG     T0     g711ulaw    VOIP        P33333      239.X.X.X:21000                   

ISR4451# show voip rtp connections
VoIP RTP Port Usage Information:
Max Ports Available: 19999, Ports Reserved: 101, Ports in Use: 0
Port range not configured
                                        Min   Max   Ports     Ports     Ports  
Media-Address Range                     Port  Port  Available Reserved  In-use 
------------------------------------------------------------------------------
Global Media Pool                       8000  48198 19999     101       0      
------------------------------------------------------------------------------
VoIP RTP active connections :
No. CallId     dstCallId  LocalRTP RmtRTP   LocalIP                                RemoteIP                                  MPSS VRF 
1     116        115        15986    21000    192.0.2.1                           239.X.X.X                               NO   NA                  
Found 1 active RTP connections

ISR4451# show voice port summary
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/3/1           --  e&m-imd     up    up   trunked  trunked  y 

ISR4451# show voice call summary
PORT           CODEC     VAD VTSP STATE            VPM STATE
============== ========= === ===================== =====================
0/3/1          g711ulaw   y  S_CONNECT             S_TRUNKED                

ISR4451# show voice call status
CallID     CID  ccVdb      Port        Slot/Bay/DSP:Ch  Called #   Codec    MLPP Dial-peers
0x73       12D0 0x7F7475CF8C08 0/3/1            0/3/1:1 33333      g711ulaw  4 777 33333777/33333
1 active call found

ISR4451# show voice trunk-conditioning supervisory 
FAST SCAN
0/3/1 : state : TRUNK_SC_CONN_DEFAULT_OOS, voice : off , signal : on ,master
status: lost keepalive, trunk connected
sequence oos : idle and oos
pattern :rx_idle = 0000 rx_oos = 1111 
timeout timing : idle = 0, idle_off = 0, restart = 120, standby = 0, timeout = 30
supp_all = 0, supp_voice = 0, keep_alive = 5
timer: oos_ais_timer = 46, timer = 43


ISR4451# show voice trunk-conditioning signaling 

0/3/1 : 
hardware-state ACTIVE signal type is NorthamericanCAS
status : lost keepalive,
forced playout pattern = 0xF
idle monitoring : disabled
tx_idle = FALSE, rx_idle = FALSE, tx_oos = FALSE, lost_keepalive = TRUE
trunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0,tx_oos_timer = 0

要验证IP到模拟的复制，请先检查新的IOS-XE命令：

ISR4451# show platform hardware qfp active feature sbc hootie group
SBC Hootie structure :
---------------------------------------

  VRF                                   = 0
  IP                                    = 239.X.X.X
  Port                                  = 21000
  Protocol                              = 1
  Calls in group                        = 1

SBC Hootie group Statistics
---------------------------------------

  Total RTP packets received                  = 2873
  Total RTP octects received                  = 573520
  Total RTP packets replicated                = 2873
  Total RTP octects replicated                = 573520
  Total RTP packets dropped                   = 0
  Total RTP octects dropped                   = 0


ISR4451# show platform hardware qfp active feature sbc hootie group
SBC Hootie structure :
---------------------------------------

  VRF                                   = 0
  IP                                    = 239.X.X.X
  Port                                  = 21000
  Protocol                              = 1
  Calls in group                        = 1

SBC Hootie group Statistics
---------------------------------------

  Total RTP packets received                  = 3111
  Total RTP octects received                  = 621032
  Total RTP packets replicated                = 3111
  Total RTP octects replicated                = 621032
  Total RTP packets dropped                   = 0
  Total RTP octects dropped                   = 0

组播验证

验证PIM邻居：

ISR4451# show ip pim neighbor 
PIM Neighbor Table
Mode: B - Bidir Capable, DR - Designated Router, N - Default DR Priority,
      P - Proxy Capable, S - State Refresh Capable, G - GenID Capable,
      L - DR Load-balancing Capable
Neighbor          Interface                Uptime/Expires    Ver   DR
Address                                                            Prio/Mode
Y.Y.Y.Y      GigabitEthernet0/0/1     00:20:13/00:01:41 v2    1 / DR S P G

验证mroute输出是否正确：

ISR4451# show ip mroute
[snip]

(192.0.2.1, 239.X.X.X), 00:01:08/00:02:20, flags: FT
  Incoming interface: Vif1, RPF nbr 0.0.0.0
  Outgoing interface list:
    GigabitEthernet0/0/1, Forward/Sparse, 00:01:08/00:03:19

验证列表中是否包含组播RP:

ISR4451# show ip igmp member
Flags: A  - aggregate, T - tracked
       L  - Local, S - static, V - virtual, R - Reported through v3 
       I - v3lite, U - Urd, M - SSM (S,G) channel 
       1,2,3 - The version of IGMP, the group is in
Channel/Group-Flags: 
       / - Filtering entry (Exclude mode (S,G), Include mode (G))
Reporter:
       <mac-or-ip-address> - last reporter if group is not explicitly tracked
       <n>/<m>      - <n> reporter in include mode, <m> reporter in exclude

 Channel/Group                  Reporter        Uptime   Exp.  Flags  Interface 
 *,239.X.X.X                  192.0.2.2    00:01:16 01:43 2VA    Vi1

验证组播数据包复制：

RP# show ip mroute count
[snip]
Group: 239.X.X.X, Source count: 1, Packets forwarded: 2107, Packets received: 2108
  RP-tree: Forwarding: 2/0/56/0, Other: 2/0/0
  Source: 192.168.19.1/32, Forwarding: 2105/50/158/80, Other: 2106/0/1


RP# show ip mroute count
[snip]
Group: 239.X.X.X, Source count: 1, Packets forwarded: 2190, Packets received: 2191
  RP-tree: Forwarding: 2/0/56/0, Other: 2/0/0
  Source: 192.168.19.1/32, Forwarding: 2188/50/159/80, Other: 2189/0/1

思科 CLI 分析器（仅适用于注册客户）支持某些 show 命令。要查看对 show 命令输出的分析，请使用思科 CLI 分析器。

故障排除

本部分提供了可用于对配置进行故障排除的信息。

呼叫设置问题

 如果连接未建立，请首先通过以下调试检验信令：

debug vpm signal
debug voip vtsp session
debug voip ccapi inout

调试示例：

123165: Oct  XX 13:21:55.563: htsp_process_event: [0/3/1, S_DOWN, E_HTSP_IF_INSERVICE]
123166: Oct  XX 13:21:55.564: %LINK-3-UPDOWN: Interface recEive and transMit 0/3/1, changed state to up
123167: Oct  XX 13:21:55.564: recEive and transMit 0/3/1 rx_signal_map:
 0 0 0 0
 0 0 0 0
 8 8 8 8
 8 8 8 8
123168: Oct  XX 13:21:55.564: recEive and transMit 0/3/1 tx_signal_map:
 0 0 0 0
 0 0 0 0
 C C C C
 C C C C
123169: Oct  XX 13:21:55.564: htsp_process_event: [0/3/1, S_OPEN_PEND, E_HTSP_GO_TRUNK]em_trunk_null_init 
123170: Oct  XX 13:21:55.564: flex_set_Legerity_impedance: [0/3/1] impedance = 0
123171: Oct  XX 13:21:55.704: htsp_process_event: [0/3/1, S_TRUNK_NULL, E_HTSP_INSERVE]default_trunk_down
123172: Oct  XX 13:21:55.704: htsp_timer - 6204 msec
123173: Oct  XX 13:21:55.919: %SYS-5-CONFIG_I: Configured from console by vty3 (192.168.19.2)
123174: Oct  XX 13:22:01.908: htsp_process_event: [0/3/1, S_TRUNK_PEND, E_HTSP_EVENT_TIMER]
123175: Oct  XX 13:22:01.908: htsp_timer_stop htsp_setup_ind
123176: Oct  XX 13:22:01.908: [0/3/1] get_local_station_id calling num= calling name= calling time=10/08 13:22  orig called=
123177: Oct  XX 13:22:01.908: htsp_timer - 2000 msec

123181: Oct  XX 13:22:01.909: //-1/80F08D0180E8/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x7F7475CF8C08, Call Info(
   Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=33333(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=777, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1

123203: Oct  XX 13:22:01.911: //115/80F08D0180E8/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=33333(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=80F08D01-CA55-11E8-80E8-8E0AC3C8E4C4, Outgoing Dial-peer=33333

123252: Oct  XX 13:22:01.914: //116/80F08D0180E8/CCAPI/cc_api_caps_ack:
   Destination Interface=0x7F7475CF8C08, Destination Call Id=115, Source Call Id=116,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=AGGRESSIVE(0x4),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=2165)
123253: Oct  XX 13:22:01.914: //115/80F08D0180E8/CCAPI/cc_api_caps_ack:
   Destination Interface=0x7F7471175B68, Destination Call Id=116, Source Call Id=115,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=AGGRESSIVE(0x4),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=2165)


123255: Oct  XX 13:22:01.914: //115/80F08D0180E8/VTSP:(0/3/1):-1:1:1/vtsp_call_connect: Connected Name 
123256: Oct  XX 13:22:01.914: //115/80F08D0180E8/VTSP:(0/3/1):-1:1:1/vtsp_call_connect: Connected Number 33333
123257: Oct  XX 13:22:01.914: //115/80F08D0180E8/VTSP:(0/3/1):-1:1:1/vtsp_call_connect: Connected oct3a  0
123258: Oct  XX 13:22:01.914: //115/80F08D0180E8/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)

123265: Oct  XX 13:22:01.916: htsp_process_event: [0/3/1, S_TRUNK_W_CUTTHRU, E_HTSP_VOICE_CUT_THROUGH]
123266: Oct  XX 13:22:01.916: send_trunk_dsp_voice_chnl_mapping:[0/3/1], 1/0/0 
123267: Oct  XX 13:22:01.916: send_trunk_dsp_sig_chnl_mapping:[0/3/1], 129/0/0 
123268: Oct  XX 13:22:01.916: recEive and transMit 0/3/1 rx_signal_map:
 0 0 0 0
 0 0 0 0
 0 0 0 0
 0 0 0 8 default_trunk_up
123269: Oct  XX 13:22:01.916: recEive and transMit 0/3/1 tx_signal_map:
 0 0 0 0
 0 0 0 0
 F F F F
 F F F F default_trunk_updefault_trunk_up
123270: Oct  XX 13:22:01.916: recEive and transMit 0/3/1 rx_signal_map:
 0 0 0 0
 0 0 0 0
 0 0 0 0
 0 0 0 8 default_trunk_up
123271: Oct  XX 13:22:01.916: recEive and transMit 0/3/1 tx_signal_map:
 0 0 0 0
 0 0 0 0
 F F F F
 F F F F default_trunk_up
123272: Oct XX 13:22:01.916: %HTSP-5-UPDOWN: Trunk port(channel) [0/3/1] is up

如果看到此错误，则是由于拨号对等体上的session protocol multicast命令不可用。

%VOICE_IEC-3-GW: H323: Internal Error (H225 chn, sock fail in RAS): IEC=1.1.186.5.81.0

音频问题

如果问题出在无音频上，请验证语音网关是否已正确加入组播组。有关工作设备的基线输出，请参阅本文档验证部分中的命令输出。特定组播组的show ip mroute命令的传出接口决不能为Null。如果看到为Null的传出接口，请检查组播LAN的适用网络配置，因为这表明语音网关无法正确加入组播组。

Null传出接口示例：

Router# show ip mroute 239.X.X.X
(*, 239.X.X.X), 00:22:02/stopped, RP 10.188.0.1, flags: SJCF
  Incoming interface: GigabitEthernet0/0/1, RPF nbr X.X.X.X
  Outgoing interface list:
    Vif1, Forward/Sparse-Dense, 00:18:27/00:02:32

(A.B.C.D, 239.X.X.X), 00:20:34/00:01:23, flags: PFT
  Incoming interface: Vif1, RPF nbr 0.0.0.0
  Outgoing interface list: Null 

如果设备在组播组中正确但音频问题依然存在，请使用命令show platform hardware qfp active feature sbc hootie group多次验证设备是否可以接收和复制数据包。每次运行该命令时，计数器必须增加。或者，可以运行show platform hardware qfp active statistics drop命令以查看语音网关是否丢弃流量。要清除这些计数器，请运行show platform hardware qfp active statistics drop clear命令。

如果未配置IP multicast-routing，则Ipv4mcNoRoute增量的丢弃原因如下所示：

4451# show platform hardware qfp active statistics drop
-------------------------------------------------------------------------
Global Drop Stats                         Packets                  Octets
-------------------------------------------------------------------------
Ipv4mcNoRoute                                 728                  145272

其他音频问题（例如网关无法将从模拟端收到的组播RTP数据包复制到IP端的问题）可能会因组播配置问题而发生。当观察到这些丢包时，这些问题可能表现为丢包原因FIAError。观察这些情况时，查看适用的组播配置，确保网关可以正确加入组播组，并且show ip mroute命令具有有效的输出接口。请参阅本文档的多播部分，了解基线命令输出。

4451# show platform hardware qfp active statistics drop
-------------------------------------------------------------------------
Global Drop Stats                         Packets                  Octets
-------------------------------------------------------------------------
FIAError                                      724                  144800

如果未启用组播路由，则显示ip mroute states的输出。

ISR4451# sh ip mroute
IP Multicast Forwarding is not enabled.
[snip]

PCM捕获

为了验证语音端口上是否发送或接收了模拟音频，您可以捕获PCM。完整的PCM文档

conf t
voice pcm capture buffer 200000
voice pcm capture destination bootflash:
exit
!
test voice port 0/1/0 pcm-dump caplog ffffff duration 255
! send audio
test voice port 0/1/0 pcm-dump disable
!
copy flash:/<filename>.dat [ftp://user:pass@ip.address/filename.pcap | tftp://a.b.c.d/filename]
! TAC is required to decode the binary .dat file into SIN/SOUT/RIN audio streams

数据包捕获(PCAP)

为了验证是否发送或接收了组播RTP，您可以在物理接口上进行数据包捕获(PCAP)。完整的EPC文档。

! NOT IN CONFIGURATION TERMINAL
monitor capture TAC int gig0/0/1 both
monitor capture TAC match any
!
monitor capture TAC start
! send audio
monitor capture TAC stop
!
monitor capture TAC export [flash:/filename.pcap | ftp://user:pass@ip.address/filename.pcap | tftp://a.b.c.d/filename]
!
monitor capture TAC clear

DSP测试音

如果需要，测试音可由语音网关上的DSP/PVDM在所需方向（网络 — IP-LAN端或本地模拟端口端）生成。

此音调可以定向到DSP以指向IP LAN组播地址。这些命令可用于启用/禁用。连接必须处于活动状态，并且您必须指定用于测试的模拟端口。

test voice port 0/1/0 inject-tone network 1000
! A 1000hz tone is now being generated from the analog port to the IP LAN Multicast Address
test voice port 0/1/0 inject-tone network disable

为了从DSP到模拟端口外生成声音，可以使用这些命令启用/禁用。连接必须处于活动状态，并且您必须指定用于测试的模拟端口。

test voice port 0/1/0 inject-tone local 1000
! A 1000hz tone is now being generated out of the analog port.
test voice port 0/1/0 inject-tone local disable

使用VLC媒体播放器测试组播接收

下载VLC Media Player并导航到Media > Open Network Stream

以此格式输入组播RTP IP地址并按播放

rtp://@239.X.X.X:21000

然后下载并打开Wireshark。然后选择数据包捕获所需的特定接口。

使用rtp过滤器开始捕获。

如果一切顺利，您必须加入组播RP。（可以从RP运行相同的组播命令以验证加入组播组的PC）。

通过tone命令生成音频或让模拟终端发言。

现在，您必须在wireshark中看到数据包。请记住，源IP必须是VIF IP减去1，因此我们的测试必须是192.0.2.2 -1 = 192.0.2.1。

思科 CLI 分析器（仅适用于注册客户）支持某些 show 命令。要查看对 show 命令输出的分析，请使用思科 CLI 分析器。

注意：使用 debug 命令之前，请参阅有关 Debug 命令的重要信息。

相关信息

• 已知缺陷

CSCvd18792 - ISR4K - Hoot和Holler E&M端口不能与组播集线器共置

CSCve66876 - ISR4K — 丢弃来自DSP的数据包的组播RP注册

CSCve71893 - ISR4K - Hoot和Holler组播复制问题

• 技术支持和文档 — Cisco System