SIP:Connection-Oriented Media (Comedia) Enhancements for SIP [Cisco IOS Software Releases 12.2 T]

Table Of Contents

SIP: Connection-Oriented Media Enhancements for SIP

Contents

Prerequisites for Connection-Oriented Media (Comedia) Enhancements for SIP

Restrictions for Connection-Oriented Media (Comedia) Enhancements for SIP

Information About Connection-Oriented Media (Comedia) Enhancements for SIP

Benefits of Connection-Oriented Media (Comedia) Enhancements for SIP

Symmetric NAT Traversal

Sample SDP Message

Symmetric NAT Call Flows

How to Configure Comedia Enhancements for SIP

Configuring Connection-Oriented Media (Comedia) Enhancements for SIP

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP

Configuration Examples for Connection-Oriented Media (Comedia) Enhancements for SIP

Configuring Source Media Check Example

Configuring the Endpoint Connection Role Example

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP Examples

Additional References

Related Documents

Standards

MIBs

RFCs

Technical Assistance

Command Reference

nat symmetric check media-src

nat symmetric role

Glossary

SIP: Connection-Oriented Media Enhancements for SIP

The Connection-Oriented Media (Comedia) Enhancements for SIP feature allows the Cisco gateway to check the media source of incoming Realtime Transport Protocol (RTP) packets, and allows the endpoint to advertise its presence inside or outside of Network Address Translation (NAT). Using the new feature enables symmetric NAT traversal by supporting the capability to modify and update an existing RTP session remote address and port.

Feature Specifications for Connection-Oriented Media (Comedia) Enhancements for SIP

Feature History

Release

Modification

12.2(13)T

This feature was introduced.

Supported Platforms

Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5850, and Cisco CVA 120 series platforms.

Availability of Cisco IOS Software Images

Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.

Contents

•Prerequisites for Connection-Oriented Media (Comedia) Enhancements for SIP

•Restrictions for Connection-Oriented Media (Comedia) Enhancements for SIP

•Information About Connection-Oriented Media (Comedia) Enhancements for SIP

•How to Configure Comedia Enhancements for SIP

•Configuration Examples for Connection-Oriented Media (Comedia) Enhancements for SIP

•Additional References

•Command Reference

•Glossary

Prerequisites for Connection-Oriented Media (Comedia) Enhancements for SIP

•Ensure that your Cisco router has the minimum memory requirements necessary for voice capabilities.

•Ensure that the gateway has voice functionality configured for SIP.

For more information about configuring SIP, refer to

Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2, the "Configuring SIP for VoIP" chapter.

•Establish a working IP network.

For more information about configuring IP, refer to

Cisco IOS IP Configuration Guide, Release 12.2.

•Configure NAT.

For more information about configuring NAT, refer to:

Configuring Network Address Translation: Getting Started.

Restrictions for Connection-Oriented Media (Comedia) Enhancements for SIP

The new feature does not support the a=direction:both attribute of the Session Description Protocol (SDP) message, as defined in the Internet Engineering Task Force (IETF) draft, draft-ietf-mmusic-sdp-comedia-04.txt, Connection-Oriented Media Transport in SDP. There is likewise no corresponding command-line interface (CLI) command. If the SIP gateway receives an SDP message specifying a=direction:both, the endpoint is treated by the gateway as active, and considered to be inside the NAT.

Note Proxy parallel forking is not supported with this feature unless all endpoints reply with 180 message response without SDP, because this feature does not handle media coming from multiple endpoints simultaneously.

Information About Connection-Oriented Media (Comedia) Enhancements for SIP

To configure the Connection-Oriented Media (Comedia) Enhancements for SIP feature, you must understand the following concepts:

•Benefits of Connection-Oriented Media (Comedia) Enhancements for SIP

•Symmetric NAT Traversal

•Sample SDP Message

•Symmetric NAT Call Flows

Benefits of Connection-Oriented Media (Comedia) Enhancements for SIP

•Ability to check the media source address and port of incoming RTP packets, thereby enabling the remote address and port of the existing session to be updated

•Enhanced interoperability in networks where NAT devices are unaware of SIP or SDP signaling

•Ability to advertise endpoint presence inside or outside NAT

•Ability to specify the connection role of the endpoint

Symmetric NAT Traversal

The Connection-Oriented Media (Comedia) Enhancements for SIP feature provides the following functionality to symmetric NAT traversal:

•Allows the Cisco gateway to check the media source of incoming (RTP) packets.

•Allows the endpoint to advertise its presence inside or outside of NAT.

NAT, which maps the source IP address of a packet from one IP address to a different IP address, has varying functionality and configurations. NAT can help conserve IP version 4 (IPv4) addresses, or it can be used for security purposes to hide the IP address and LAN structure behind the NAT. VoIP endpoints may both be outside NAT, both inside, or one inside and the other outside.

In symmetric NAT, all requests from an internal IP address and port to a specific destination IP address and port are mapped to the same external IP address and port. The new feature provides additional capabilities for symmetric NAT traversal.

Prior to the implementation of connection-oriented media enhancements, NAT traversal presented challenges for both SIP, which signals the protocol messages that set up a call, and for RTP, the media stream that transports the audio portion of a VoIP call. An endpoint with connections to clients behind NATs and on the open Internet had no way of knowing when to trust the addressing information it received in the SDP portion of SIP messages, or whether to wait until it received a packet directly from the client before opening a channel back to the source IP:port of that packet. Once a VoIP session was established, the endpoint was, in some scenarios, sending packets to an unreachable address. This scenario typically occurred in NAT networks that were SIP-unaware.

In addition to the challenges posed by NAT traversal in SIP, NAT traversal in RTP requires that a client must know what type of NAT it sits behind, and that it must also obtain the public address for an RTP stream. Any RTP connection between endpoints outside and inside NAT must be established as a point-to-point connection. The external endpoint must wait until it receives a packet from the client so that it knows where to reply. The connection-oriented protocol used to describe this type of session is known as Connection-Oriented Media (Comedia), as defined in the IETF draft, draft-ietf-mmusic-sdp-comedia-04.txt, Connection-Oriented Media Transport in SDP.

The Connection-Oriented Media (Comedia) Enhancements for SIP feature implements one of many possible SIP solutions to address problems with different NAT types and traversals. With the new feature the gateway can open an RTP session with the remote end and then update or modify the existing RTP session remote address and port (raddr:rport) with the source address and port of the actual media packet received after passing through NAT. The new feature allows you to configure the gateway to modify the RTP session remote address and port by implementing support for the SDP direction (a=direction:<role>) attribute defined in, draft-ietf-mmusic-sdp-comedia-04.txt, Connection-Oriented Media Transport in SDP. Valid values for the attribute are as follows:

•active, which indicates that the endpoint initiates a connection to the port number on the m= line of the session description from the other endpoint.

•passive, which indicates that the endpoint accepts a connection to the port number on the m= line of the session description sent from itself to the other endpoint.

•both, which indicates the that endpoint both accepts an incoming connection and initiates an outgoing connection to the port number on the m= line of the session description from the other endpoint.

The new feature introduces new CLI commands to configure the following SIP user agent settings for symmetric NAT:

•The nat symmetric check-media-src command enables checking the incoming packet for media source address. This capability allows the gateway to check the source address and update the media session with the new remote media address and port.

•The nat symmetric role command specifies the function of the endpoint in the connection setup procedure. The role keyword may be set to one of the following:

–active, meaning the endpoint initiates a connection to the port number on the m= line of the session description from the other endpoint.

–passive, meaning the endpoint accepts a connection to the port number on the m= line of the session description sent from itself to the other endpoint.

Note The Cisco comedia implementation does not support a=direction:both. If the Cisco gateway receives a=direction:both in the SDP message, the endpoint is considered active.

Sample SDP Message
v=o
o=CiscoSystemsSIP-GW-UserAgent 5732 7442 IN IP4 10.15.66.43
s=SIP Call
c=IN IP4 10.15.66.43
t=0 0
m=audio 17306 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
a=direction:passive
Symmetric NAT Call Flows

The following call flow diagrams describe call setup during symmetric NAT traversal scenarios. Figure 1 shows a NAT device that is unaware of SIP or SDP signaling. The SIP endpoints are not communicating the connection-oriented media direction role in the SDP message. The originating gateway is configured, using the command nat symmetric check-media-src, to detect the media source and update the VoIP RTP session to the network address translated address:port pair.

Figure 1 SIP Endpoints Not Communicating the SDP direction:<role> Attribute

Figure 2 shows a NAT device that is unaware of SIP or SDP signaling, but the SIP endpoints are communicating the connection-oriented media direction role in the SDP message. The originating gateway is configured as a passive entity in the network using the nat symmetric role command. When the passive entity receives a direction role of active, it updates the VoIP RTP session to the network address translated address:port pair.

Figure 2 SIP Endpoints Communicating the SDP direction:<role>

Note Configuring the originating gateway for passive or active setting can differ from the NAT device setup. The SIP user agent communicates the CLI configured direction role in the SDP body. Checking for media packets is automatically enabled only if the gateway receives a direction role of active or both.

How to Configure Comedia Enhancements for SIP

This section contains the following procedures. Each procedure is identified as either required or optional.

•Configuring Connection-Oriented Media (Comedia) Enhancements for SIP (required)

•Verifying Connection-Oriented Media (Comedia) Enhancements for SIP (optional)

Configuring Connection-Oriented Media (Comedia) Enhancements for SIP

Perform the following tasks to enable the gateway to check the media source address and port of the first incoming RTP packet, and to optionally specify whether the endpoint is active or passive. Once the media source check is enabled, the gateway can modify or update the established VoIP RTP session with upstream addressing information extracted from the SDP body of the received request.

SUMMARY STEPS

1. enable

2. configure [terminal | memory | network]

3. sip-ua

4. nat symmetric check-media-source

5. nat symmetric role {active | passive}

6. exit

DETAILED STEPS

Command or Action

Purpose

Step 1

enable
Example:

Router> enable

Enables higher privilege levels, such as privileged EXEC mode.

•Enter your password if prompted.

Step 2

configure {terminal | memory | network}
Example:

Router# configure terminal

Enters global configuration mode.

Step 3

Router(config# sip-ua
Example:

Router(config# sip-ua

Enters SIP-UA configuration mode.

Step 4

nat symmetric check-media-source
Example:

Router(config-sip-ua)# nat symmetric check-media-source

(Required) Specifies settings for the SIP user agent in symmetric NAT configuration.

•In this example, the gateway is configured to perform source media checking for symmetric NAT.

Step 5

nat symmetric role {active | passive}
Example:

Router(config-sip-ua)# nat symmetric role active

(Optional) Specifies endpoint settings for the SIP user agent in symmetric NAT configuration. The default setting is passive.

•The optional active keyword configures the endpoint ability to initiate an outgoing connection.

•The optional passive keyword configures the endpoint ability to accept an incoming connection.

Step 6

exit
Example:

Router(config-sip-ua)# exit

Exits SIP user agent configuration mode

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP

Perform this task to verify that the Connection-Oriented Media (Comedia) Enhancements for SIP feature is working.

SUMMARY STEPS

1. enable

2. show running-config

3. debug ccsip all

DETAILED STEPS

Command or Action

Purpose

Step 1

enable
Example:

Router> enable

Enables higher privilege levels, such as privileged EXEC mode.

•Enter your password if prompted.

Step 2

show running-config
Example:

Router# show running-config

(Optional) Displays the configuration information currently running on the router. View the sip-ua configuration to verify Comedia settings.

Step 3

debug ccsip all
Example:

Router# debug ccsip all

(Optional) Enables all SIP call tracing. View the direction attribute settings and port and network address translation traces to verify Comedia configuration.

Configuration Examples for Connection-Oriented Media (Comedia) Enhancements for SIP

This section provides the following configuration examples:

•Configuring Source Media Check Example

•Configuring the Endpoint Connection Role Example

•Verifying Connection-Oriented Media (Comedia) Enhancements for SIP Examples

Configuring Source Media Check Example

The following example shows how to enable checking the media source address and port of incoming RTP packets:
Router(config)# sip-ua
Router(config-sip-ua)# nat symmetric check-media-src
Configuring the Endpoint Connection Role Example

The following example shows how to configure the endpoint role in connection setup to passive:
Router(config)# sip-ua
Router(config-sip-ua)# nat symmetric role passive

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP Examples

In the following examples, the output is displayed for each command used in the section "Verifying Connection-Oriented Media (Comedia) Enhancements for SIP."

Sample Output for the show running-config Command

You can use the show running-config command to verify that source media checking is enabled:
Router# show running-config
Building configuration...
Current configuration :2791 bytes
!
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname 3640
!
voice-card 2
!
ip subnet-zero
!
no ip domain lookup
ip domain name cisco.com
ip name-server 172.18.195.113
!
isdn switch-type primary-ni
!
fax interface-type fax-mail
mta receive maximum-recipients 0
ccm-manager mgcp
!
controller T1 2/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
!
controller T1 2/1
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
!
interface Ethernet0/0
 ip address 172.18.197.22 255.255.255.0
 half-duplex
!
interface Serial0/0
 no ip address
 shutdown
!
interface TokenRing0/0
 no ip address
 shutdown
 ring-speed 16
!
interface FastEthernet1/0
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial2/0:23
 no ip address
 no logging event link-status
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
interface Serial2/1:23
 no ip address
 no logging event link-status
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn outgoing display-ie
 no cdp enable
!
ip classless
ip route 0.0.0.0 0.0.0.0 Ethernet0/0
no ip http server
ip pim bidir-enable
!
call rsvp-sync
!
voice-port 2/0:23
!
voice-port 2/1:23
!
voice-port 3/0/0
!
voice-port 3/0/1
!
mgcp ip qos dscp cs5 media
mgcp ip qos dscp cs3 signaling
!
mgcp profile default
!
dial-peer cor custom
!
dial-peer voice 646 voip
destination-pattern 5552222
 session protocol sipv2
 session target ipv4:10.0.0.1
!
dial-peer voice 700 pots
 destination-pattern 700#T
port 0:D
!
gateway 
!
sip-ua 
nat symmetric check-media-src
max-forwards 5
!
line con 0
line aux 0
line vty 0 4
 login
!
end
Sample Output for the debug ccsip all Command

In the following example, output is displayed with the role keyword of the nat symmetric command set to active for the originating gateway, and to passive for the terminating gateway.
Router3640# debug ccsip all
All SIP call tracing enabled
Router3640#
00:02:12:0x6327E424 :State change from (UNDEFINED, SUBSTATE_NONE)  to (STATE_IDLE, 
SUBSTATE_NONE)
00:02:12:****Adding to UAC table
00:02:12:adding call id 3 to table
00:02:12:Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP (10)
00:02:12:CCSIP-SPI-CONTROL: act_idle_call_setup
00:02:12: act_idle_call_setup:Not using Voice Class Codec
00:02:12:act_idle_call_setup:preferred_codec set[0] type :g711ulaw bytes:160 
00:02:12:sipSPICopyPeerDataToCCB:From CLI:Modem NSE payload = 100, Passthrough = 0,Modem 
relay = 0, Gw-Xid = 1
SPRT latency 200, SPRT Retries = 12, Dict Size = 1024
 String Len = 32, Compress dir = 3
00:02:12:****Deleting from UAC table
00:02:12:****Adding to UAC table
00:02:12:Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION (6)
00:02:12:0x6327E424 :State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_IDLE, 
SUBSTATE_CONNECTING)
00:02:12:0x6327E424 :State change from (STATE_IDLE, SUBSTATE_CONNECTING)  to (STATE_IDLE, 
SUBSTATE_CONNECTING)
00:02:12:sipSPIUsetBillingProfile:sipCallId for billing records = 
D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
00:02:12:CCSIP-SPI-CONTROL: act_idle_connection_created
00:02:12:CCSIP-SPI-CONTROL: act_idle_connection_created:Connid(1) created to 
172.18.200.237:5060, local_port 56992
00:02:12:CCSIP-SPI-CONTROL: sipSPIOutgoingCallSDP
00:02:12: Preferred method of dtmf relay is:6, with payload :101
00:02:12: convert_codec_bytes_to_ptime:Values :Codec:g711ulaw codecbytes :160, ptime:20
00:02:12:sip_generate_sdp_xcaps_list:Modem Relay disabled. X-cap not needed
00:02:12:CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT:Using GMT
00:02:12:sipSPIAddLocalContact
00:02:12:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
00:02:12:sip_stats_method
00:02:12:0x6327E424 :State change from (STATE_IDLE, SUBSTATE_CONNECTING)  to 
(STATE_SENT_INVITE, SUBSTATE_NONE)
00:02:12:Sent:
INVITE sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Supported:timer,100rel
Min-SE: 1800
Cisco-Guid:3563045876-351146444-2147852364-2382746380
User-Agent:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Max-Forwards:1
Timestamp:730944132
Contact:<sip:888001@10.15.66.43:5060;user=phone>
Expires:60
Allow-Events:telephone-event
Content-Type:application/sdp
Content-Length:291
v=0
o=CiscoSystemsSIP-GW-UserAgent 9502 9606 IN IP4 10.15.66.43
s=SIP Call
c=IN IP4 10.15.66.43
t=0 0
m=audio 16398 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=direction:active
00:02:12:CCSIP-SPI-CONTROL: act_sentinvite_wait_100
00:02:12:CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT:Using GMT
00:02:12:sipSPIAddLocalContact
00:02:12:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
00:02:12:sip_stats_method
00:02:12:Sent:
INVITE sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Supported:timer,100rel
Min-SE: 1800
Cisco-Guid:3563045876-351146444-2147852364-2382746380
User-Agent:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Max-Forwards:1
Timestamp:730944132
Contact:<sip:888001@10.15.66.43:5060;user=phone>
Expires:60
Allow-Events:telephone-event
Content-Type:application/sdp
Content-Length:291
v=0
o=CiscoSystemsSIP-GW-UserAgent 9502 9606 IN IP4 10.15.66.43
s=SIP Call
c=IN IP4 10.15.66.43
t=0 0
m=audio 16398 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=direction:active
00:02:12:Received:
SIP/2.0 100 Trying
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:57:53 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Timestamp:730944132
Server:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Allow-Events:telephone-event
Content-Length:0
00:02:12:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:172.18.200.237:5060
00:02:12:CCSIP-SPI-CONTROL: act_sentinvite_new_message
00:02:12:CCSIP-SPI-CONTROL: sipSPICheckResponse
00:02:12:sip_stats_status_code
00:02:12: Roundtrip delay 32 milliseconds for method INVITE
00:02:12:0x6327E424 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE)  to 
(STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
00:02:13:Received:
SIP/2.0 183 Session Progress
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:57:53 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Timestamp:730944132
Server:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Require:100rel
RSeq:5975
Allow-Events:telephone-event
Contact:<sip:2021010124@172.18.200.237:5060;user=phone>
Content-Type:application/sdp
Content-Disposition:session;handling=required
Content-Length:240
v=0
o=CiscoSystemsSIP-GW-UserAgent 1692 40 IN IP4 172.18.200.237
s=SIP Call
c=IN IP4 172.18.200.237
t=0 0
m=audio 16898 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
a=direction:passive
00:02:13:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:172.18.200.237:5060
00:02:13:CCSIP-SPI-CONTROL: act_recdproc_new_message
00:02:13:CCSIP-SPI-CONTROL: sipSPICheckResponse
00:02:13:sip_stats_status_code
00:02:13: Roundtrip delay 708 milliseconds for method INVITE
00:02:13:sipSPIGetSdpBody :Parse incoming session description
00:02:13:HandleSIP1xxSessionProgress:Content-Disposition received in 18x 
response:session;handling=required
00:02:13:sipSPIDoFaxMediaNegotiation()
00:02:13:sipSPIDoMediaNegotiation:Codec (g711ulaw) Negotiation Successful on Static 
Payload
00:02:13: sipSPIDoPtimeNegotiation:One ptime attribute found - value:20
00:02:13: convert_ptime_to_codec_bytes:Values :Codec:g711ulaw ptime :20, codecbytes:160
00:02:13: convert_codec_bytes_to_ptime:Values :Codec:g711ulaw codecbytes :160, ptime:20
00:02:13: Parsed the direction:role identified as:0
00:02:13:sipSPIDoDTMFRelayNegotiation:Requested DTMF-RELAY option(s) not found in 
Preferred DTMF-RELAY option list!
00:02:13: sipSPIDoMediaNegotiation:DTMF Relay mode :Inband Voice
00:02:13:sip_sdp_get_modem_relay_cap_params:
00:02:13:sip_sdp_get_modem_relay_cap_params:NSE payload from X-cap = 0
00:02:13:sip_do_nse_negotiation:NSE Payload 100 found in SDP
00:02:13:sip_do_nse_negotiation:Remote NSE payload = local one = 100, Use it
00:02:13:sip_select_modem_relay_params:X-tmr not present in SDP. Disable modem relay
00:02:13:sipSPIDoQoSNegotiation - SDP body with media description
00:02:13:sipSPIUpdCcbWithSdpInfo:SDP Media Information:
Negotiated Codec      :g711ulaw , bytes :160
Early Media           :0 
Delayed Media         :0 
Bridge Done           :0 
New Media             :0 
DSP DNLD Reqd         :0 
Media Dest addr/Port  :172.18.200.237:16898 
Orig Media Addr/Port  :0.0.0.0:0 
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROCEEDING)  to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROGRESS)
00:02:13:ccsip_process_response_contact_record_route
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROGRESS)  to (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)
00:02:13:Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION (6)
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)  to 
(STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)
00:02:13:sipSPIRtcpUpdates:rtcp_session info
                laddr = 10.15.66.43, lport = 16398, raddr = 172.18.200.237, rport=16898
00:02:13:sipSPIRtcpUpdates:NO extraction of source address from remote media 
00:02:13: sipSPIRtcpUpdates No rtp session in bridge, create a new one
00:02:13:CCSIP-SPI-CONTROL: ccsip_caps_ind
00:02:13:ccsip_get_rtcp_session_parameters:CURRENT VALUES:
ccCallID=3, current_seq_num=0x1500
00:02:13:ccsip_get_rtcp_session_parameters:NEW VALUES:
ccCallID=3, current_seq_num=0xB93
00:02:13:ccsip_caps_ind:Load DSP with negotiated codec :g711ulaw, Bytes=160
00:02:13:sipSPISetDTMFRelayMode:set DSP for dtmf-relay = 
CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB
00:02:13:sip_set_modem_caps:Negotiation already Done. Set negotiated Modem caps
00:02:13:sip_set_modem_caps:Modem Relay & Passthru both disabled
00:02:13:sip_set_modem_caps:nse payload = 100, ptru mode = 0, ptru-codec=0, redundancy=0, 
xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
00:02:13:ccsip_caps_ind:Load DSP with codec :g711ulaw, Bytes=160
00:02:13:CCSIP-SPI-CONTROL: ccsip_caps_ack
00:02:13:CCSIP-SPI-CONTROL: act_recdproc_connection_created
00:02:13:CCSIP-SPI-CONTROL: sipSPICheckSocketConnection:Connid(2) created to 
172.18.200.237:5060, local_port 50689
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_CONNECTING)  to 
(STATE_RECD_PROCEEDING, SUBSTATE_NONE)
00:02:13:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
00:02:13:sip_stats_method
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE)  to 
(STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROGRESS)
00:02:13:Received:
SIP/2.0 200 OK
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:57:53 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Timestamp:730944132
Server:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Allow-Events:telephone-event
Contact:<sip:2021010124@172.18.200.237:5060;user=phone>
Content-Type:application/sdp
Content-Length:240
v=0
o=CiscoSystemsSIP-GW-UserAgent 1692 40 IN IP4 172.18.200.237
s=SIP Call
c=IN IP4 172.18.200.237
t=0 0
m=audio 16898 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
a=direction:passive
00:02:13:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:172.18.200.237:5060
00:02:13:Sent:
PRACK sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
CSeq:102 PRACK
RAck:5975 101 INVITE
Content-Length:0
00:02:13:CCSIP-SPI-CONTROL: act_recdproc_new_message
00:02:13:CCSIP-SPI-CONTROL: sipSPICheckResponse
00:02:13:sip_stats_status_code
00:02:13: Roundtrip delay 736 milliseconds for method PRACK
00:02:13:sipSPIGetSdpBody :Parse incoming session description
00:02:13:CCSIP-SPI-CONTROL: sipSPIUACSessionTimer
00:02:13:CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing
00:02:13:CCSIP-SPI-CONTROL: act_recdproc_continue_200_processing:*** This ccb is the 
parent
00:02:13:sipSPIDoFaxMediaNegotiation()
00:02:13:sipSPIDoMediaNegotiation:Codec (g711ulaw) Negotiation Successful on Static 
Payload
00:02:13: sipSPIDoPtimeNegotiation:One ptime attribute found - value:20
00:02:13: convert_ptime_to_codec_bytes:Values :Codec:g711ulaw ptime :20, codecbytes:160
00:02:13: convert_codec_bytes_to_ptime:Values :Codec:g711ulaw codecbytes :160, ptime:20
00:02:13: Parsed the direction:role identified as:0
00:02:13:sipSPIDoDTMFRelayNegotiation:Requested DTMF-RELAY option(s) not found in 
Preferred DTMF-RELAY option list!
00:02:13: sipSPIDoMediaNegotiation:DTMF Relay mode :Inband Voice
00:02:13:sip_sdp_get_modem_relay_cap_params:
00:02:13:sip_sdp_get_modem_relay_cap_params:NSE payload from X-cap = 0
00:02:13:sip_do_nse_negotiation:NSE Payload 100 found in SDP
00:02:13:sip_do_nse_negotiation:Remote NSE payload = local one = 100, Use it
00:02:13:sip_select_modem_relay_params:X-tmr not present in SDP. Disable modem relay
00:02:13: sipSPICompareSDP:Flags set:NEW_MEDIA :0 DSPDNLD REQD:0 
00:02:13:sipSPIUpdCcbWithSdpInfo Bridge was done and there are no fqdn queries in 
progress, do RTCP updates
00:02:13:sipSPIRtcpUpdates:rtcp_session info
                laddr = 10.15.66.43, lport = 16398, raddr = 172.18.200.237, rport=16898
00:02:13:sipSPIRtcpUpdates:NO extraction of source address from remote media 
00:02:13: sipSPIRtcpUpdates rtp session already created in bridge - update
00:02:13:sipSPIUpdCcbWithSdpInfo:SDP Media Information:
Negotiated Codec      :g711ulaw , bytes :160
Early Media           :0 
Delayed Media         :0 
Bridge Done           :1048576 
New Media             :0 
DSP DNLD Reqd         :0 
Media Dest addr/Port  :172.18.200.237:16898 
Orig Media Addr/Port  :0.0.0.0:0 
00:02:13:sipSPIProcessMediaChanges
00:02:13:ccsip_process_response_contact_record_route
00:02:13:CCSIP-SPI-CONTROL: sipSPIProcess200OKforinvite
00:02:13:RequestCloseConnection:Closing connid 1 Local Port 50689
00:02:13:Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION (8)
00:02:13:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
00:02:13:sip_stats_method
00:02:13:0x6327E424 :State change from (STATE_RECD_PROCEEDING, 
SUBSTATE_PROCEEDING_PROGRESS)  to (STATE_ACTIVE, SUBSTATE_NONE)
00:02:13:The Call Setup Information is :
Call Control Block (CCB) :0x6327E424
State of The Call        :STATE_ACTIVE
TCP Sockets Used         :NO
Calling Number           :888001
Called Number            :2021010124
Negotiated Codec         :g711ulaw
Negotiated Codec Bytes   :160
Negotiated Dtmf-relay    :0
Dtmf-relay Payload       :0
00:02:13:
 Source IP Address (Sig  ):10.15.66.43
Source IP Address (Media):10.15.66.43
Source IP Port    (Media):16398
Destn  IP Address (Media):172.18.200.237
Destn  IP Port    (Media):16898
Destn SIP Req Addr:Port  :172.18.200.237:5060
Destn SIP Resp Addr:Port :0.0.0.0:0
Destination Name         :172.18.200.237
00:02:13:
 Orig Destn IP Address:Port (Media):0.0.0.0:0
00:02:13:udpsock_close_connect:Socket fd:1 closed for connid 1 with remote port:5060
00:02:13:Sent:
ACK sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Max-Forwards:1
Content-Length:0
CSeq:101 ACK
00:02:13:Received:
SIP/2.0 200 OK
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:57:54 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Server:Cisco-SIPGateway/IOS-12.x
CSeq:102 PRACK
Content-Length:0
00:02:13:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:172.18.200.237:5060
00:02:13:CCSIP-SPI-CONTROL: act_active_new_message
00:02:13:CCSIP-SPI-CONTROL: sact_active_new_message_response
00:02:13:CCSIP-SPI-CONTROL: sipSPICheckResponse
00:02:27:Queued event from SIP SPI :SIPSPI_EV_CC_CALL_DISCONNECT (15)
00:02:27:CCSIP-SPI-CONTROL: act_active_disconnect
00:02:27:RequestCloseConnection:Closing connid 2 Local Port 50689
00:02:27:Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION (8)
00:02:27:Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION (6)
00:02:27:0x6327E424 :State change from (STATE_ACTIVE, SUBSTATE_NONE)  to (STATE_ACTIVE, 
SUBSTATE_CONNECTING)
00:02:27:0x6327E424 :State change from (STATE_ACTIVE, SUBSTATE_CONNECTING)  to 
(STATE_ACTIVE, SUBSTATE_CONNECTING)
00:02:27:udpsock_close_connect:Socket fd:2 closed for connid 2 with remote port:5060
00:02:27:CCSIP-SPI-CONTROL: sipSPICheckSocketConnection:Connid(1) created to 
172.18.200.237:5060, local_port 54607
00:02:27:0x6327E424 :State change from (STATE_ACTIVE, SUBSTATE_CONNECTING)  to 
(STATE_ACTIVE, SUBSTATE_NONE)
00:02:27:CCSIP-SPI-CONTROL: act_active_connection_created Call Disconnect - Sending Bye
00:02:27:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
00:02:27:sip_stats_method
00:02:27:0x6327E424 :State change from (STATE_ACTIVE, SUBSTATE_NONE)  to 
(STATE_DISCONNECTING, SUBSTATE_NONE)
00:02:27:Sent:
BYE sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
User-Agent:Cisco-SIPGateway/IOS-12.x
Max-Forwards:1
Timestamp:730944147
CSeq:103 BYE
Content-Length:0
00:02:27:Received:
SIP/2.0 200 OK
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:58:08 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Server:Cisco-SIPGateway/IOS-12.x
Timestamp:730944147
Content-Length:0
CSeq:103 BYE
00:02:27:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:172.18.200.237:5060
00:02:27:CCSIP-SPI-CONTROL: act_disconnecting_new_message
00:02:27:CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response
00:02:27:CCSIP-SPI-CONTROL: sipSPICheckResponse
00:02:27:sip_stats_status_code
00:02:27: Roundtrip delay 16 milliseconds for method BYE
00:02:27:CCSIP-SPI-CONTROL: sipSPICallCleanup
00:02:27:sipSPIIcpifUpdate :CallState:4 Playout:0 DiscTime:14742 ConnTime 13360
00:02:27:0x6327E424 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to 
(STATE_DEAD, SUBSTATE_NONE)
00:02:27:The Call Setup Information is :
Call Control Block (CCB) :0x6327E424
State of The Call        :STATE_DEAD
TCP Sockets Used         :NO
Calling Number           :888001
Called Number            :2021010124
Negotiated Codec         :g711ulaw
Negotiated Codec Bytes   :160
Negotiated Dtmf-relay    :0
Dtmf-relay Payload       :0
00:02:27:
 Source IP Address (Sig  ):10.15.66.43
Source IP Address (Media):10.15.66.43
Source IP Port    (Media):16398
Destn  IP Address (Media):172.18.200.237
Destn  IP Port    (Media):16898
Destn SIP Req Addr:Port  :172.18.200.237:5060
Destn SIP Resp Addr:Port :0.0.0.0:0
Destination Name         :172.18.200.237
00:02:27:
 Orig Destn IP Address:Port (Media):0.0.0.0:0
00:02:27:
 Disconnect Cause (CC)    :16
Disconnect Cause (SIP)   :200
00:02:27:****Deleting from UAC table
00:02:27:Removing call id 3
00:02:27:RequestCloseConnection:Closing connid 1 Local Port 54607
00:02:27:Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION (8)
00:02:27: freeing ccb 6327E424
00:02:27:udpsock_close_connect:Socket fd:1 closed for connid 1 with remote port:5060
Router-3640#
Router-5300# debug ccsip all
All SIP call tracing enabled
Router-5300#
3d04h:Received:
INVITE sip:2021010124@172.18.200.237:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>
Date:Mon, 01 Mar 1993 00:02:12 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Supported:timer,100rel
Min-SE: 1800
Cisco-Guid:3563045876-351146444-2147852364-2382746380
User-Agent:Cisco-SIPGateway/IOS-12.x
CSeq:101 INVITE
Max-Forwards:1
Timestamp:730944132
Contact:<sip:888001@10.15.66.43:5060;user=phone>
Expires:60
Allow-Events:telephone-event
Content-Type:application/sdp
Content-Length:291
v=0
o=CiscoSystemsSIP-GW-UserAgent 9502 9606 IN IP4 10.15.66.43
s=SIP Call
c=IN IP4 10.15.66.43
t=0 0
m=audio 16398 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=direction:active
3d04h:HandleUdp
SocketReads :Msg enqueued for SPI with IPaddr:172.18.200.238:10105
3d04h:CCSIP-SPI-CONTROL: sipSPISipIncomingMsg
3d04h:0x629748AC :State change from (UNDEFINED, SUBSTATE_NONE)  to (STATE_IDLE, 
SUBSTATE_NONE)
3d04h:CCSIP-SPI-CONTROL: act_idle_new_message
3d04h:CCSIP-SPI-CONTROL: Clock Time Zone is UTC, same as GMT:Using GMT
3d04h:CCSIP-SPI-CONTROL: sact_idle_new_message_invite
3d04h:sip_stats_method
3d04h:CCSIP-SPI-CONTROL: sipSPIUASSessionTimer
3d04h:sipSPIGetSdpBody :Parse incoming session description
CCSIP-SPI-CONTROL: (4294967295) Warning:No network type specified in comediadir attribute.
3d04h:****Deleting from UAS Request table
3d04h:sipSPIUdeleteCcbFromTable:Entry not found for search key
3d04h:CCSIP-SPI-CONTROL: sipSPIMatchSrcIpGroup
3d04h:CCSIP-SPI-CONTROL: sipSPIContinueNewMsgInvite
3d04h:sipSPIContinueNewMsgInvite:non dial peer leg - using RTP Supported Codecs
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 18
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 0
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 8
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 4
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 2
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 15
3d04h:sipSPIContinueNewMsgInvite:RTP Preferred Codecs supported by GW 3
3d04h:sipSPIDoFaxMediaNegotiation()
3d04h:sipSPIDoMediaNegotiation:Codec (g711ulaw) Negotiation Successful on Static Payload
3d04h: sipSPIDoPtimeNegotiation:One ptime attribute found - value:20
3d04h: convert_ptime_to_codec_bytes:Values :Codec:g711ulaw ptime :20, codecbytes:160
3d04h: convert_codec_bytes_to_ptime:Values :Codec:g711ulaw codecbytes :160, ptime:20
3d04h: Parsed the SDP for direction:Extraction of src address triggered with role as :1 
3d04h:sipSPIDoDTMFRelayNegotiation:Requested DTMF-RELAY option(s) not found in Preferred 
DTMF-RELAY option list!
3d04h: sipSPIDoMediaNegotiation:DTMF Relay mode :Inband Voice
3d04h:sip_sdp_get_modem_relay_cap_params:
3d04h:sip_sdp_get_modem_relay_cap_params:NSE payload from X-cap = 0
3d04h:sip_do_nse_negotiation:NSE Payload 100 found in SDP
3d04h:sip_do_nse_negotiation:Remote NSE payload = local one = 100, Use it
3d04h:sip_select_modem_relay_params:X-tmr not present in SDP. Disable modem relay
3d04h:sipSPIUpdCcbWithSdpInfo:SDP Media Information:
Negotiated Codec      :g711ulaw , bytes :160
Early Media           :0 
Delayed Media         :0 
Bridge Done           :0 
New Media             :0 
DSP DNLD Reqd         :0 
Media Dest addr/Port  :10.15.66.43:16398 
Orig Media Addr/Port  :0.0.0.0:0 
3d04h:sipSPIHandleInviteMedia:
Negotiated Codec        :g711ulaw, bytes :160
Preferred Codec         :g729r8, bytes :20
Preferred  DTMF relay 1 :0
Preferred  DTMF relay 2 :0
Negotiated DTMF relay   :0
Preferred and Negotiated NTE payloads:101 0
Preferred and Negotiated NSE payloads:100 100
Preferred and Negotiated Modem Relay:0 0
Preferred and Negotiated Modem Relay GwXid:1 0
3d04h:sipSPIContinueNewMsgInvite:Requires reliable-provisional support
3d04h:sipSPIDoQoSNegotiation - SDP body with media description
3d04h:sipSPIAddBillingInfoToCcb:sipCallId for billing records = 
D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
3d04h:****Adding to UAS Request table
3d04h:adding call id 31 to table
3d04h:Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE (7)
3d04h:sip_stats_status_code
3d04h:****Adding to UAS Response table
3d04h:Previous Hop 10.15.66.43:5060
3d04h:0x629748AC :State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, 
SUBSTATE_NONE)
3d04h:Sent:
SIP/2.0 100 Trying
Via:SIP/2.0/UDP  10.15.66.43:5060
From:"888001" <sip:888001@10.15.66.43>;tag=20694-C53
To:<sip:2021010124@172.18.200.237;user=phone>;tag=1069B954-25F
Date:Tue, 04 Jan 2000 23:57:53 GMT
Call-ID:D6EB9E87-14EE11CC-8008A04C-8E05D30C@10.15.66.43
Timestamp:730944132
Server:Cisco-SIPGateway/IOS-12.x
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Cisco IOS Software Releases 12.2 T

SIP:Connection-Oriented Media (Comedia) Enhancements for SIP

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Table Of Contents

SIP: Connection-Oriented Media Enhancements for SIP

Contents

Prerequisites for Connection-Oriented Media (Comedia) Enhancements for SIP

Restrictions for Connection-Oriented Media (Comedia) Enhancements for SIP

Information About Connection-Oriented Media (Comedia) Enhancements for SIP

Benefits of Connection-Oriented Media (Comedia) Enhancements for SIP

Symmetric NAT Traversal

Sample SDP Message

Symmetric NAT Call Flows

How to Configure Comedia Enhancements for SIP

Configuring Connection-Oriented Media (Comedia) Enhancements for SIP

SUMMARY STEPS

DETAILED STEPS

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP

SUMMARY STEPS

DETAILED STEPS

Configuration Examples for Connection-Oriented Media (Comedia) Enhancements for SIP

Configuring Source Media Check Example

Configuring the Endpoint Connection Role Example

Verifying Connection-Oriented Media (Comedia) Enhancements for SIP Examples