Document ID: 105624
Protocol translation and repair is a key Cisco Unified Border Element (CUBE) function. CUBE can be deployed between two devices that support the same VoIP protocol (SIP), but do not interwork because of differences in how the protocol is implemented or interpreted. CUBE can customize the SIP messaging on either side to what the devices in that segment of the network expect to see by normalizing the SIP messaging on the network border, or between two non-interoperable devices within the network.
Service providers may have policies for which SIP messaging fields should be present (or what constitutes valid values for the header fields) before a SIP call enters their network. Similarly, enterprises and small businesses may have policies for the information that can enter or exit their networks for policy or security reasons from a service provider SIP trunk.
In order to customize SIP messaging in both directions, you can place CUBE with a SIP normalization configuration at the boundary of these networks as shown in this image:
In addition to network policy compliance, the CUBE SIP normalization capabilities can be used to resolve incompatibilities between SIP devices inside the enterprise network. These are the situations in which incompatibilities can arise:
A device rejects an unknown header (value or parameter) instead of ignoring it
A device sends incorrect data in a SIP message
A device does not implement (or implements incorrectly) protocol procedures
A device expects an optional header value or parameter, or an optional protocol procedure that can be implemented in multiple ways
A device sends a value or parameter that must be changed or suppressed before it leaves or enters the network
Variations in the SIP standards on how to achieve certain functions
The SIP profiles feature on CUBE provides a solution to these SIP normalization and customization issues.
Ensure that you meet these requirements before you attempt this configuration:
Basic knowledge of how to configure and use Cisco IOS voice features (such as dial-peers)
Basic knowledge of how to configure and use the Cisco Unified Border Element (CUBE)
Intermediate knowledge of the SIP protocol and its messages, fields, and parameters
The information in this document is based on these software and hardware versions:
Cisco Unified Border Element release on a Cisco ISR, Cisco AS5400XM Access Gateway, Cisco AS5350XM Access Gateway, Cisco 7200 router, or a Cisco 7300 router that runs Cisco IOS release 12.4.15XZ or later
Cisco TDM-to-SIP gateway on a Cisco ISR Cisco AS5400XM Access Gateway, or a Cisco AS5350XM Access Gateway that runs Cisco IOS release 12.4.15XZ or later
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
Refer to the Cisco Technical Tips Conventions for more information on document conventions.
The configuration of the SIP profiles feature allows you to add, modify, or remove any SIP or SDP header value in an outgoing SIP message on CUBE. A list of the exact messages and headers supported is provided in the Supported SIP Messages section of this document. SIP profiles can be configured either at the dial-peer level or the global level.
The syntax for message modification uses regular expression notation to match and replace fields in messages. Matched substrings can be used in replace patterns. When multiple regular expression rules apply to the same [method/response]:header combination, the second rule applies to the result string of the first rule.
SIP Profiles does not allow you to remove or add mandatory SIP headers. Only the modify option is available for mandatory headers. Mandatory SIP headers include To, From, Via, CSeq, Call-Id, and Max-Forwards. Mandatory SDP headers include v, o, s, t ,c, and m.
The ANY special keyword is provided in the CLI to indicate that a rule must be applied to any message within the specified category. The rules configured for an INVITE message are applied only to the first INVITE in the protocol sequence for the call. A special REINVITE keyword is supported to define operation needed on subsequent INVITEs in a protocol sequence for the call.
SIP profiles can also be used to change a header name from the long form to the compact form; for example, From to f. This can be used as a way to reduce the length of a SIP message. By default Cisco IOS SIP never sends the compact form of the SIP messages although it receives either the long or the short form.
The SIP profiles feature affects only outgoing SIP messages. The rules are applied as the last step before the message leaves the CUBE router; that is, after destination dial-peer matching has taken place. Changes to the SIP messages are not remembered or acted on by the CUBE application.
The Content-length field is recalculated after the SIP Profiles rules are applied to the outgoing message.
The general command that defines a rule to add a field to a SIP method/response is:
<request/response> <message code> <sip-header/sdp-header> <header-name> add <add-value>
The general command that defines a rule to remove a field to a SIP method/response is:
<request/response> <message code> <sip-header/sdp-header> <header-name> remove
The general command that defines a rule to modify a field to a SIP method/response is:
<request/response> <message code> <sip-header/sdp-header> <header-name> modify <match-pattern> <replace-pattern>
The first step is to define the rules. In order to define the rules, use the general command structure given in the previous section. For example:
voice class sip-profiles 100 request INVITE sip-header… response 100 sip-header… request INVITE sdp-header…
The second step is to apply the rules either to the global or dial-peer level of the CUBE configuration. In order to apply the rules globally to all calls traversing CUBE, use this command structure:
voice service voip sip sip-profiles 100
In order to apply the rules selectively to calls traversing only a particular outgoing dial-peer, use this command structure:
dial-peer voice 555 voip voice-class sip-profiles 100
If rules are configured at both the global and dial-peer level, the dial-peer configuration takes precedence over global level configuration.
These notes apply generically to all Cisco IOS features that use regular expressions (which includes SIP profiles):
When an add-value, match-pattern or replace-pattern contain white-space characters, the entire value must be included between double quotes. For example:
response 100 sip-header add “User-Agent: CISCO CUBE”
When an add-value, match-pattern or replace-pattern contains double quotes ( " ) and white space characters, a backslash ( \ ) must prefix the inner quotes. For example, in order to add "CISCO" CUBE, use this command:
response 100 sip-header add “User-Agent: \”CISCO\” CUBE”
In order to provide the most flexibility, syntax checking is not performed on the SIP messages that result after the rules are applied. You must ensure through adequate testing that the changes you specify in the profile rules result in valid SIP protocol exchanges.
The SIP message type cannot be changed with SIP profiles.That is, a 180 Ringing response cannot be changed to a 183 Session Progress response.
Mandatory headers can only be modified; they cannot be added or removed. Mandatory SIP headers include To, From, Via, CSeq, Call-Id, and Max-Forwards. Mandatory SDP headers include v, o, s, t ,c, and m.
While regular expression variables can be used in match and replace substrings in order to store and reuse values, information cannot be extracted from one message and applied to another. For example, the calling number cannot be extracted from the INVITE and inserted into a subsequent REFER message. The rules specified apply to a single message at a time; only information in that message is manipulated.
This section provides examples of SIP profile rules to achieve specific changes to SIP messages.
This section provides examples of how to add SIP and SDP headers to messages.
Action: Add the b=AS:4000 SDP header to the video-media line
voice class sip-profiles 100 request INVITE sdp-header Video-Bandwidth-Info add "b=AS:4000"
Message: 480 Temporarily Not Available
Action: Add the Retry-After SIP header
voice class sip-profiles 100 response 480 sip-header Retry-After add “Retry-After: 60”
Message: INVITEs and REINVITEs
Action: Add the “user=phone” tag to the SIP URI header
voice class sip-profiles 100 request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
Message: 200 Response
Action: Add the User-Agent SIP header
voice class sip-profiles 100 response 200 sip-header User-Agent add "User-Agent: CiscoSystems-SIP-GW-UA"
This section provides examples on how to remove SIP and SDP headers to messages.
Message: All requests and responses
Action: Remove the Cisco-Guid SIP header
voice class sip-profiles 100 request ANY sip-header Cisco-Guid remove response ANY sip-header Cisco-Guid remove
Message: BYE and CANCEL
Action: Remove the Reason SIP header
voice class sip-profiles 100 request BYE sip-header Reason remove request CANCEL sip-header Reason remove
Message: 100 and 180 Responses
Action: Remove the Server SIP header
voice class sip-profiles 100 response 100 sip-header Server remove response 180 sip-header Server remove
This section provides examples on how to modify SIP and SDP headers in messages.
Action: Modify the From: header to “gateway@gw-ip-address” format e.g. change firstname.lastname@example.org to email@example.com
voice class sip-profiles 100 request INVITE sip-header From modify "(<.*:)(.*@)" "\1gateway@"
Action: replace “CiscoSystems-SIP-GW-UserAgent” with “-” in the o= line of the SDP header
voice class sip-profiles 100 request INVITE sdp-header Session-Owner modify "CiscoSystems-SIP-GW-UserAgent“ "-"
Action: Convert “sip url” to “tel url” in the Req-URI, From and To headers e.g. from “sip:firstname.lastname@example.org:5060” to “tel:2222000020”
voice class sip-profiles 100 request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1" request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>" request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"
This section provides a full configuration example for a CUBE router configured with SIP profiles.
CUBE#show run Building configuration... Current configuration : 5888 bytes ! ! Last configuration change at 13:16:50 CDT Mon Feb 11 2008 ! NVRAM config last updated at 13:24:35 CDT Mon Feb 11 2008 ! version 12.4 service timestamps debug datetime msec localtime service timestamps log datetime msec localtime service password-encryption ! hostname CUBE ! boot-start-marker boot system flash:c2800nm-ipvoice_ivs-mz.124-18.2.2.PIA1p.bin boot-end-marker ! logging message-counter syslog logging buffered 2000000 no logging console no logging monitor enable lab 0 ! no aaa new-model memory-size iomem 10 clock timezone CDT -6 clock summer-time CDT recurring no network-clock-participate slot 1 ! voice-card 0 no dspfarm ! ip cef ! voice service voip media flow-around allow-connections sip to sip sip sip-profiles 100 ! voice class sip-profiles 100 request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" ! interface GigabitEthernet0/0 ip address x.x.x.x 255.255.255.0 duplex full speed 100 ! ip http server ! control-plane ! dial-peer voice 100 voip destination-pattern .T session protocol sipv2 session target ipv4:x.x.x.x dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua ! line con 0 line aux 0 line vty 0 4 exec-timeout 90 0 ! scheduler allocate 20000 1000 end
The header length (which includes the header name) should not exceed 300 characters after modification with SIP profiles. The maximum header length for an add value is approximately 220 characters. The maximum SDP header length is 2048 characters. If any header length exceeds the maximum value after a rule is applied, that rule will be ignored, and the changes are not applied. If the total SDP length exceeds 2048 characters after modifications, all changes to the SDP are ignored and not applied.
The SIP profiles feature cannot be used to drop an entire SIP message; it can only be used to manipulate (add, modify, or remove) the content in the message
Regular expression variables can be used to extract and store parameters from an existing header, but the values stored in variables in one rule can be used ONLY in the replace-pattern of the same rule; it can NOT be used by any other rules, which means you cannot insert the values extracted by one rule into another rule.
Content specified between open parenthesis ( ( ) and closed parenthesis ( ) ) in a match-pattern are stored in variables denoted by 1, 2, 3, ... 9 in the order that they are found. The stored values of these variables can then be inserted again in the replace-pattern by referencing the variables with \1, \2, ...\9 respectively.
For example, if we want to change:
Remote-Party-ID: “CUBE” <sip:email@example.com>;privacy=off;screen=no to P-Asserted-Identity: “CUBE” <sip:firstname.lastname@example.org>
This can be achieved by the following SIP Profile rule:
request INVITE sip-header Remote-Party-ID modify "Remote-Party-ID:(.*>).*" "P-Asserted-Identity:\1"
In this example, the sequence (.*>) matches “CUBE” <sip:email@example.com>. This value is stored in variable 1, which is referenced in the replace-pattern with \1.
This section provides the CLI options of the SIP messages that can be customized with the CUBE SIP profiles feature.
These SIP requests are supported:
router(config-class)#request ? ACK sip ack ANY any sip request BYE sip bye CANCEL sip cancel COMET sip comet INFO sip info INVITE sip invite NOTIFY sip notify OPTIONS sip options PRACK sip prack PUBLISH sip publish REFER sip refer REGISTER sip register REINVITE sip reinvite SUBSCRIBE sip subscribe UPDATE sip info
These SIP responses are supported:
router(config-class)#response ? 100 Response code 100 180 Response code 180 181 Response code 181 182 Response code 182 183 Response code 183 200 Response code 200 202 Response code 202 300 Response code 300 301 Response code 301 302 Response code 302 305 Response code 305 380 Response code 380 400 Response code 400 401 Response code 401 402 Response code 402 403 Response code 403 404 Response code 404 405 Response code 405 406 Response code 406 407 Response code 407 408 Response code 408 409 Response code 409 410 Response code 410 412 Response code 412 413 Response code 413 414 Response code 414 415 Response code 415 416 Response code 416 417 Response code 417 420 Response code 420 421 Response code 421 422 Response code 422 423 Response code 423 480 Response code 480 481 Response code 481 482 Response code 482 483 Response code 483 484 Response code 484 485 Response code 485 486 Response code 486 487 Response code 487 488 Response code 488 489 Response code 489 491 Response code 491 493 Response code 493 500 Response code 500 501 Response code 501 502 Response code 502 503 Response code 503 504 Response code 504 505 Response code 505 513 Response code 513 580 Response code 580 600 Response code 600 603 Response code 603 604 Response code 604 606 Response code 606 ANY Any Response
These SIP headers are supported:
rtr(config-class)#request INVITE sip-header ? Accept-Contact Accept-Encoding Accept-Header Accept-Language Accept-Resource-Priority Alert-Info Allow-Events Allow-Header Also Authorization CC-Diversion CC-Redirect CSeq Call-ID Call-Info Cisco-Gcid Cisco-Guid Contact Content-Disposition Content-Encoding Content-Id Content-Length Content-Type Date Diversion Event Expires From History-Info Location MIME-Version Max-Forwards Min-Expires Min-SE Orig-dial-plan P-Asserted-Identity P-Preferred-Identity Privacy Proxy-Authenticate Proxy-Authorization Proxy-Require Rack Reason Record-Route Refer-To Referred-By Reject-Contact Remote-Party-ID Replaces Request-Disposition Requested-By Require Resource-Priority Retry-After Route Rseq SIP-ETag SIP-If-Match SIP-Req-URI Server Session-Expires Session-Header Subscription-State Supported Term-dial-plan Timestamp To Unsupported User-Agent Via WWW-Authenticate Warning
These SDP headers are supported:
rtr(config-class)#response 200 sdp-header ? Attribute a= Audio-Attribute a= Audio-Bandwidth-Info b= Audio-Connection-Info c= Audio-Encryption-Key k= Audio-Media m=audio Audio-Session-Info i= Bandwidth-Key b= Connection-Info c= Email-Address e= Encrypt-Key k= Phone-Number p= Repeat-Times r= Session-Info i= Session-Name s= Session-Owner o= Time-Adjust-Key z= Time-Header t= Url-Descriptor u= Version v= Video-Attribute a= Video-Bandwidth-Info b= Video-Connection-Info c= Video-Encryption-Key k= Video-Media m=video Video-Session-Info i=
The INVITE messages with (and without) a SIP profiles configuration applied are shown in this section. Use this method to verify that the SIP profile rules in the configuration affect the correct and desired changes in SIP messages.
This example shows a sample configuration:
voice class sip-profiles 1 request INVITE sdp-header Audio-Bandwidth-Info add "b=AS:1600“ request ANY sip-header Cisco-Guid remove request INVITE sdp-header Session-Owner modify "CiscoSystems-SIP-GW-UserAgent" "-“
This example shows a sample SIP INVITE message without the SIP profiles configuration applied (salient fields are highlighted in bold):
INVITE sip:firstname.lastname@example.org:5060 SIP/2.0 Via: SIP/2.0/UDP 18.104.22.168:5060;branch=z9hG4bK1A203F From: "sipp " <sip:email@example.com>;tag=F11AE0-1D8D To: <sip:firstname.lastname@example.org> Date: Mon, 29 Oct 2007 19:02:04 GMT Call-ID: 4561B116-858811DC-804DEF2E-4CF2D71B@22.214.171.124 Cisco-Guid: 1163870326-2240287196-2152197934-1290983195 Content-Length: 290 v=0 o=CiscoSystemsSIP-GW-UserAgent 6906 8069 IN IP4 126.96.36.199 s=SIP Call c=IN IP4 188.8.131.52 t=0 0 m=audio 17070 RTP/AVP 0 c=IN IP4 184.108.40.206 a=rtpmap:0 PCMU/8000 a=ptime:20
This example shows the same sample SIP INVITE message with the SIP profiles configuration applied (changed fields are highlighted in bold):
INVITE sip:email@example.com:5060 SIP/2.0 Via: SIP/2.0/UDP 220.127.116.11:5060;branch=z9hG4bK1A203F From: "sipp " <sip:firstname.lastname@example.org>;tag=F11AE0-1D8D To: <sip:email@example.com> Date: Mon, 29 Oct 2007 19:02:04 GMT Call-ID: 4561B116-858811DC-804DEF2E-4CF2D71B@18.104.22.168 Content-Length: 279 v=0 o=- 6906 8069 IN IP4 22.214.171.124 s=SIP Call c=IN IP4 126.96.36.199 t=0 0 m=audio 17070 RTP/AVP 0 c=IN IP4 188.8.131.52 a=rtpmap:0 PCMU/8000 a=ptime:20 b=AS:1600
debug ccsip all is a useful troubleshooting command for SIP profiles.
In this example, the text “voice class SIP Profiles” shows which SIP profile is applied. Here is a sample of the command output:
router#debug ccsip all … Oct 12 06:51:53.619: //-1/735085DC8F3D/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : sippOct 12 06:51:53.619: //-1/735085DC8F3D/SIP/Info/sipSPIGetCallConfig: Peer tag 2 matched for incoming call Oct 12 06:51:53.619: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetCallConfig: voice class SIP Profiles tag is set : 1 Oct 12 06:51:53.619: //-1/735085DC8F3D/SIP/Info/sipSPIGetCallConfig: Not using Voice Class Codec Oct 12 06:51:53.619: //-1/735085DC8F3D/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled Oct 12 06:51:53.619: //-1/735085DC8F3D/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH …
In this example, the text “sip_profiles” highlights the modifications performed by the SIP profiles configuration. Here is a sample of the command output:
router#debug ccsip all … Oct 12 06:51:53.647: //-1/xxxxxxxxxxxx/SIP/Info/ sip_profiles_application_change_sdp_line: New SDP header is added : b=AS: 1600 Oct 12 06:51:53.647: //-1/xxxxxxxxxxxx/SIP/Info/ sip_profiles_update_content_length: Content length header before modification : Content-Length: 290 Oct 12 06:51:53.647: //-1/xxxxxxxxxxxx/SIP/Info/ sip_profiles_update_content_length: Content length header after modification : Content-Length: 279 …
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|Updated: Apr 22, 2008||Document ID: 105624|