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This document explains how Cisco Unified SIP Proxy (CUSP) call routing logic.
Contributed by Joshua Meadows, Cisco TAC Engineer.
Cisco recommends that you have the knowledge of these topics:
Term |
Definition |
Network |
A SIP network is a logical collection of local interfaces that can be treated the same for general routing purposes. The Network logically defines areas of the network. The network can be defined using interfaces on the CUSP device, or specific ports can be used to provide segmentation. To achieve this logical segmentation, separate Listen Ports can be configured. (Example: Listen Ports 14.50.245.9:5060, 14.50.245.9:5062, 14.50.245.9:5065 can define three logical networks using a single CUSP layer 3 interface) Once the Networks are defined logically, they can be used to configured Triggers based on the Network. Note: If you set up a listen port, ensure devices sending traffic to the CUSP use the correct port. If you set up listen port 14.50.245.9:5065 for CUCM traffic, you must ensure CUCM sends traffic to port 5065, not the default of 5060. |
Triggers |
Triggers can be set to identify incoming messages. Triggers can identify Inbound Network, Local Port, Remote Network, etc. |
Server Group |
Server groups define the elements with which the Cisco Unified SIP Proxy system interacts for each network. Both Server Group and Route Group can be used as destinations in the Route Table. A server group would generally be used for redundant devices of the same type. A CUBE stack would be a good example of a Server Group. |
Route Group |
A route group allows you to designate the order in which gateways and trunks are selected. It allows you to prioritize a list of gateways and ports for outgoing trunk selection. Both Server Group and Route Group can be used as destinations in the Route Table. A route group generally defines a weighted group destinations to reach the same device. A direct SIP trunk to a CUCM and a SIP trunk to a PSTN gateway to reach the CUCM would be a good example of a Route Group. The direct SIP trunk to the CUCM would be the prefered method, and the PSTN route would be a backup. |
Route Table |
You configure route tables to direct SIP requests to their appropriate destinations. Each route table consists of a set of keys that are matched based on the lookup policy. Route Tables in CUSP are similar to Layer 3 routing tables. CUSP Route Tables consist of Keys similar to networks in layer 3 routing tables. Route Tables link Keys to destinations. In the CUSP Route Table Keys can be mapped to the following route types to route SIP messages: destination: a specific host or a locally configured server group can be configured as a destination route-group: a locally configured route-group with one or more elements route-policy: route policies can be used to move between Route Tables similar to translation patterns in CUCM reponse: rather than routing a SIP message, CUSP can send a specific response to terminate the call attempt default-sip: Simple routing following RFC 3263. Note: If mapping a Key to a route-policy, be cognizant of logical loops. |
Route Policy |
A Route Policy points to a a Route Table and defines how to use the Key in that route table. Example: Route Table Name: "FromCUCM105-RT" Lookup Key matches: "Prefix-Longest-Match" Lookup Key: "SIP Header: 'To' Phone" By separating the definition of the Key from the configured value of the Key the same Route Table can be used in different ways. For example, one Route Policy could define the Route Table's Key as the prefix for a TO: header, while another Route Policy could define the Route Table's Key as the prefix for a FROM: header. |
Routing Triggers |
Routing Triggers link a Trigger to a Route Policy. Logically it states if a SIP message matches the Trigger, then use the configured Route Policy. |
In summation, a SIP message is tagged with a Network based on the SIP listen port. The Network can be used to match a Trigger. The Route Policy then identifies which Route Table to use based on the Trigger and defines where to look for the Key. The Route Table will then use the Key to find out where to route the SIP message (Route Type). The Route Type (Host, Server Group, Route Group, etc) will be used to send the SIP message to the configured destination (element).
Call from PSTN 1001 to 2003 on CUCM115
Incoming Network: “PSTN”
Trigger: “From-PSTN-Trigger”
Triggers if incoming message matched Network “PSTN”
Routing Trigger: “FromPSTN-RPolicy” “From-PSTN-Trigger”
Links “From-PSTN-Trigger” to “FromPSTN-RPolicy”
Route Policy: “FromPSTN-RPolicy”
Specifies Routing table “PSTN-RT”
Specifies Lookup Key Matches “Prefix-Longest-Match”
Specifies Lookup Key is “SIP Header: 'To' Phone”
Route Table: “PSTN-RT”
Contains Key “2” to go to Route Group “CUCM115_RG”
Route Group (or Server Group): “CUCM115_RG”
Contains Element 14.50.245.20:5065
These configurations combine to make the logical statement:
For a call from the PSTN, where phone number prefix is 2, route to 14.50.245.20:5065
PSTN - 2XXX and 5XXX calls are sent to CUSP via the CUBE and vCUBE
CUCM 10.5 - 1XXX and 2XXX are sent to CUSP via SIP trunk
CUCM 11.5 - 1XXX and 5XXX are sent to CUSP via SIP trunk
Note: When using the GUI, some configurations must be committed before they are available in other configuration sections. These are marked with ###Commit Configuration
CLI Configuration | GUI Configuration |
Create a Network | |
sip network PSTN standard |
Configure >> Networks >> Add
|
Define listening port to identify network 'PSTN' | |
sip listen PSTN udp 14.50.245.9 5060 |
Configure >> Networks >> [Network Name] >> SIP Listen Points >> Add
|
Trigger for Inbound Network 'PSTN' | |
trigger condition From-PSTN-Trigger sequence 1 in-network ^\QPSTN\E$ end sequence end trigger condition |
Configure >> Triggers >> Add Configure Trigger Name Configure Trigger Condition and click add |
Specify a destination for 'CUCM115_RG' | |
route group CUCM115_RG element target-destination 14.50.245.20:5065:udp CUCM115 q-value 0.0 failover-codes 502 - 503 weight 50 end element end route |
Configure >> Route Groups >> Add (###Commit Configuration) Configure a Route Group Name Click "Click Here" under Elements Column, then Click Add Enter the Element Destination |
Define Route Table and associate a key to a destination | |
route table PSTN-RT key 2 group CUCM115_RG key 5 group CUCM105_RG end route table |
Configure >> Route Tables >> Add (###Commit Configuration)
Enter a Key and Destination
When configuring a Route Group as a destination in a Route Table, do NOT add a port and transport type. By adding a Port and/or Transport type, you are telling CUSP to look for DNS Host entry Cubestack:5060:UDP rather than looking in the locally significant Server Group configurations.
|
Define the Key for 'FromPSTN-RPolicy' | |
policy lookup FromPSTN-RPolicy sequence 100 PSTN-RT header to uri-component phone rule prefix end sequence end policy |
Configure >> Route Policies >> Add (###Commit Configuration) Configure a Route Policy name
Click Add to add a Policy Step
The policy step will define how the Key is used. In this case, the policy looks for the longest Phone number match on the To: field in the SIP header |
Link the 'From-PSTN-Trigger' to 'FromPSTN-RPolicy' | |
trigger routing sequence 2 policy FromPSTN-RPolicy condition From-PSTN-Trigger |
Configure >> Routing Triggers >> Add Select a Routing Policy to link to a Trigger
|
Note: show configuration active verbose will show the entire configuration including the Route Tables.
josmeado-CUSP(cusp)# show configuration active verbose Building CUSP configuration... ! server-group sip global-load-balance weight server-group sip retry-after 250 server-group sip element-retries udp 2 server-group sip element-retries tls 1 server-group sip element-retries tcp 1 sip dns-srv enable no naptr end dns ! no sip header-compaction no sip logging ! sip max-forwards 70 sip network CUCM105 standard no non-invite-provisional allow-connections no tls verify retransmit-count invite-client-transaction 3 retransmit-count invite-server-transaction 5 retransmit-count non-invite-client-transaction 3 retransmit-timer T1 500 retransmit-timer T2 4000 retransmit-timer T4 5000 retransmit-timer TU1 5000 retransmit-timer TU2 32000 retransmit-timer clientTn 64000 retransmit-timer serverTn 64000 tcp connection-setup-timeout 1000 tls handshake-timeout 3000 udp max-datagram-size 1500 end network ! sip network CUCM115 standard no non-invite-provisional allow-connections no tls verify retransmit-count invite-client-transaction 3 retransmit-count invite-server-transaction 5 retransmit-count non-invite-client-transaction 3 retransmit-timer T1 500 retransmit-timer T2 4000 retransmit-timer T4 5000 retransmit-timer TU1 5000 retransmit-timer TU2 32000 retransmit-timer clientTn 64000 retransmit-timer serverTn 64000 tcp connection-setup-timeout 1000 tls handshake-timeout 3000 udp max-datagram-size 1500 end network ! sip network PSTN standard no non-invite-provisional allow-connections no tls verify retransmit-count invite-client-transaction 3 retransmit-count invite-server-transaction 5 retransmit-count non-invite-client-transaction 3 retransmit-timer T1 500 retransmit-timer T2 4000 retransmit-timer T4 5000 retransmit-timer TU1 5000 retransmit-timer TU2 32000 retransmit-timer clientTn 64000 retransmit-timer serverTn 64000 tcp connection-setup-timeout 1000 tls handshake-timeout 3000 udp max-datagram-size 1500 end network ! sip overload reject retry-after 0 ! no sip peg-counting ! sip privacy service sip queue message drop-policy head low-threshold 80 size 2000 thread-count 20 end queue ! sip queue radius drop-policy head low-threshold 80 size 2000 thread-count 20 end queue ! sip queue request drop-policy head low-threshold 80 size 2000 thread-count 20 end queue ! sip queue response drop-policy head low-threshold 80 size 2000 thread-count 20 end queue ! sip queue st-callback drop-policy head low-threshold 80 size 2000 thread-count 10 end queue ! sip queue timer drop-policy none low-threshold 80 size 2500 thread-count 8 end queue ! sip queue xcl drop-policy head low-threshold 80 size 2000 thread-count 2 end queue ! route recursion ! sip tcp connection-timeout 30 sip tcp max-connections 256 ! no sip tls ! sip tls connection-setup-timeout 1 ! trigger condition From-CUCM105-Trigger sequence 1 in-network ^\QCUCM105\E$ end sequence end trigger condition ! trigger condition From-CUCM115-Trigger sequence 1 in-network ^\QCUCM115\E$ end sequence end trigger condition ! trigger condition From-PSTN-Trigger sequence 1 in-network ^\QPSTN\E$ end sequence end trigger condition ! trigger condition mid-dialog sequence 1 mid-dialog end sequence end trigger condition ! accounting no enable no client-side no server-side end accounting ! server-group sip group Cubestack PSTN element ip-address 14.50.245.6 5060 udp q-value 0.0 weight 1 element ip-address 14.50.245.7 5060 udp q-value 0.0 weight 1 failover-resp-codes 503 lbtype weight ping end server-group ! route group CUCM105_RG element target-destination 14.50.245.25:5062:udp CUCM105 q-value 0.0 failover-codes 510 weight 50 end element end route ! route group CUCM115_RG element target-destination 14.50.245.20:5065:udp CUCM115 q-value 0.0 failover-codes 502 - 503 weight 50 end element end route ! route table FromCUCM105-RT key * target-destination Cubestack PSTN key 2 group CUCM115_RG end route table ! route table FromCUCM115-RT key 1 target-destination Cubestack PSTN key 5 group CUCM105_RG end route table ! route table PSTN-RT key 2 group CUCM115_RG key 5 group CUCM105_RG end route table ! policy lookup FromCUCM105-RPolicy sequence 100 FromCUCM105-RT header to uri-component phone rule prefix end sequence end policy ! policy lookup FromCUCM115-RPolicy sequence 100 FromCUCM115-RT header to uri-component phone rule prefix end sequence end policy ! policy lookup FromPSTN-RPolicy sequence 100 PSTN-RT header to uri-component phone rule prefix end sequence end policy ! trigger routing sequence 1 by-pass condition mid-dialog trigger routing sequence 2 policy FromPSTN-RPolicy condition From-PSTN-Trigger trigger routing sequence 3 policy FromCUCM115-RPolicy condition From-CUCM115-Trigger trigger routing sequence 4 policy FromCUCM105-RPolicy condition From-CUCM105-Trigger ! server-group sip global-ping ! no server-group sip ping-503 ! sip cac session-timeout 720 sip cac PSTN 14.50.245.6 5060 udp limit -1 sip cac PSTN 14.50.245.7 5060 udp limit -1 ! no sip cac ! sip listen CUCM105 udp 14.50.245.9 5062 sip listen CUCM115 udp 14.50.245.9 5065 sip listen PSTN udp 14.50.245.9 5060 ! call-rate-limit 100 ! end
In the CUSP GUI, navigate to Troubleshoot >> Cisco Unified SIP Proxy >> Traces
Trigger-Conditions - Level:debug: This will show which triggers were match to initate call routing.
Routing - Level:debug: This will show what was done during call routing. Which Key's were matched, what destination was chosen, etc.
SIP-Wire-Log - Level:debug: This will show the SIP messages received and sent.
Via GUI
In the CUSP GUI, navigate to Troubleshoot >> Cisco Unified SIP Proxy >> Traces
Select Download Log File
You can also Clear Logs
Via FTP Client
By default there is no account with FTP privilages. To enable an account with FTP privilages add the user to a PFS group.
josmeado-CUSP# user platformadmin group ? Administrators System administrators group pfs-privusers PFS privileged users group pfs-readonly PFS read only group josmeado-CUSP# user platformadmin group pfs
Via FTP Client, connect to CUSP. File Path: cusp >> log >> trace >> trace.log
13:24:36:987 08:17:2017 vCUSP,9.1.5,josmeado-CUSP,14.50.245.9,trace.log [REQUESTI.7] DEBUG 2017.08.17 13:25:03:006 conditions.RegexCondition - inNetwork='PSTN' [REQUESTI.7] DEBUG 2017.08.17 13:25:03:006 conditions.RegexCondition - IN_NETWORK: PSTN [REQUESTI.7] DEBUG 2017.08.17 13:25:03:006 conditions.AbstractRegexCondition - pattern(^\QPSTN\E$), toMatch(PSTN) returning true [REQUESTI.7] DEBUG 2017.08.17 13:25:03:006 triggers.ModuleTrigger - ModuleTrigger.eval() action<FromPSTN-RPolicy> actionParameter<> [REQUESTI.7] DEBUG 2017.08.17 13:25:03:006 triggers.ModuleTrigger - ModuleTrigger.eval() got the policy, executing it ...
In the above sample, we see the network is matched as PSTN, which is used in Route Policy "FromPSTN-RPolicy".
13:29:13:453 08:17:2017 vCUSP,9.1.5,josmeado-CUSP,14.50.245.9,trace.log [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.XCLNRSShiftRoutes - Entering ShiftAlgorithms.execute() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.XCLNRSShiftRoutes - Leaving ShiftAlgorithms.execute() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 modules.XCLLookup - Entering execute() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.XCLPrefix - Entering getKeyValue() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.FieldSelector - getToUri: To header obtained - To: <sip:2003@14.50.245.9> [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.FieldSelector - getUriPart: URI - sip:2003@14.50.245.9 part 1 [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.FieldSelector - Requested field 52 [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.FieldSelector - Returning key 2003 [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 nrs.XCLPrefix - Leaving getKeyValue() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 modules.XCLLookup - table=PSTN-RT, key=2003 [REQUESTI.7] INFO 2017.08.17 13:29:33:987 modules.XCLLookup - table is PSTN-RT [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 routingtables.RoutingTable - Entering lookup() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 routingtables.RoutingTable - Looking up 2003 in table PSTN-RT with rule prefix and modifiers=none [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 routingtables.RoutingTable - Entering applyModifiers() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:987 routingtables.RoutingTable - Leaving applyModifiers(), returning 2003 [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 routingtables.RoutingTable - Leaving lookup() [REQUESTI.7] INFO 2017.08.17 13:29:33:988 nrs.XCLPrefix - NRS Routing decision is: RouteTable:PSTN-RT, RouteKey:2, RouteGroup:CUCM115_RG [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBFactory - Entering createLoadBalancer() [REQUESTI.7] INFO 2017.08.17 13:29:33:988 loadbalancer.LBFactory - lbtype is 3(call-id) [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBFactory - Leaving createLoadBalancer() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.XCLPrefix - Stored NRSAlgResult=isFound=true, isFailure=false, Response=-1, Routes=[Ruri: 14.50.245.20:5065:udp, Route: null, Network: CUCM115, q-value=0.0radvance=[502, 503]], PolicyAdvance=null [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSAlgResult - set policyAdvance as specified in route=RouteTable:PSTN-RT, RouteKey:2, RouteGroup:CUCM115_RG [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSAlgResult - no policyAdvance specified in route [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSAlgResult - set policyAdvance as specified in algorithm={lookuprule=1, lookupfield=52, lookuplenght=-1, lookuptable=PSTN-RT, sequence=100, algorithm=1} [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSAlgResult - no policyAdvance specified in algorithm [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 modules.XCLLookup - Leaving execute() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.XCLNRSShiftRoutes - Entering ShiftRoutes.execute() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBBase - Entering getServer() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBBase - Entering initializeDomains() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSRoutes - routes before applying time policies: [Ruri: 14.50.245.20:5065:udp, Route: null, Network: CUCM115, q-value=0.0radvance=[502, 503]] [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.NRSRoutes - routes after applying time policies: [Ruri: 14.50.245.20:5065:udp, Route: null, Network: CUCM115, q-value=0.0radvance=[502, 503]] [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBBase - Leaving initializeDomains() [REQUESTI.7] INFO 2017.08.17 13:29:33:988 loadbalancer.LBHashBased - list of elements in order on which load balancing is done : Ruri: 14.50.245.20:5065:udp, Route: null, Network: CUCM115, q-value=0.0radvance=[502, 503], [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBBase - Server group route-sg selected Ruri: 14.50.245.20:5065:udp, Route: null, Network: CUCM115, q-value=0.0radvance=[502, 503] [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 loadbalancer.LBBase - Leaving getServer() [REQUESTI.7] DEBUG 2017.08.17 13:29:33:988 nrs.XCLNRSShiftRoutes - Leaving ShiftRoutes.execute()
1. CUSP gets the Key value in the TO: header
2. CUSP identifies the Key as 2003
3. CUSP Looks up the Key in the Routing Table
4. CUSP matches an entry in the Routing Table and identifies destination RouteGroup:CUCM115_RG
5. CUSP applies loadbalancing within teh RouteGroup
6. CUSP identifies the specific Element in the RouteGroup to which it will send the SIP message
7. CUSP applies Time Policies if applicable
8. CUSP finalizes the Element to which it will send a SIP Message
13:48:26:669 08:17:2017 vCUSP,9.1.5,josmeado-CUSP,14.50.245.9,trace.log [DsTransportListener-2] DEBUG 2017.08.17 13:48:52:221 DsSipLlApi.Wire - Received UDP packet on 14.50.245.9:5060 ,source 14.50.245.6:50683 INVITE sip:2003@14.50.245.9:5060 SIP/2.0 Via: SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 Remote-Party-ID: <sip:1001@14.50.245.6>;party=calling;screen=no;privacy=off From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F To: <sip:2003@14.50.245.9> Date: Thu, 17 Aug 2017 13:48:52 GMT Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 0350227076-2191790567-2162465606-1670485135 User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1502992132 Contact: <sip:1001@14.50.245.6:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 69 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 266 v=0 o=CiscoSystemsSIP-GW-UserAgent 7317 4642 IN IP4 14.50.245.6 s=SIP Call c=IN IP4 14.50.245.6 t=0 0 m=audio 8266 RTP/AVP 18 127 c=IN IP4 14.50.245.6 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 --- end of packet --- [REQUESTI.7] DEBUG 2017.08.17 13:48:52:223 DsSipLlApi.Wire - Sending UDP packet on 14.50.245.9:32789, destination 14.50.245.6:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 To: <sip:2003@14.50.245.9> From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Timestamp: 1502992132 Content-Length: 0 [REQUESTI.7] DEBUG 2017.08.17 13:48:52:225 DsSipLlApi.Wire - Sending UDP packet on 14.50.245.9:32790, destination 14.50.245.20:5065 INVITE sip:2003@14.50.245.20:5065;transport=udp SIP/2.0 Via: SIP/2.0/UDP 14.50.245.9:5065;branch=z9hG4bKM3X51yKL9BEW5v0Kudc5Dw~~128 Via: SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 Max-Forwards: 68 To: <sip:2003@14.50.245.9> From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F Contact: <sip:1001@14.50.245.6:5060> Expires: 180 Remote-Party-ID: <sip:1001@14.50.245.6>;party=calling;screen=no;privacy=off Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Content-Length: 266 Date: Thu, 17 Aug 2017 13:48:52 GMT Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 0350227076-2191790567-2162465606-1670485135 User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Timestamp: 1502992132 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=CiscoSystemsSIP-GW-UserAgent 7317 4642 IN IP4 14.50.245.6 s=SIP Call c=IN IP4 14.50.245.6 t=0 0 m=audio 8266 RTP/AVP 18 127 c=IN IP4 14.50.245.6 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 [DsTransportListener-3] DEBUG 2017.08.17 13:48:52:229 DsSipLlApi.Wire - Received UDP packet on 14.50.245.9:5065 ,source 14.50.245.20:5065 SIP/2.0 100 Trying Via: SIP/2.0/UDP 14.50.245.9:5065;branch=z9hG4bKM3X51yKL9BEW5v0Kudc5Dw~~128,SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F To: <sip:2003@14.50.245.9> Date: Thu, 17 Aug 2017 17:48:52 GMT Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Allow-Events: presence Content-Length: 0 --- end of packet --- [DsTransportListener-3] DEBUG 2017.08.17 13:48:52:284 DsSipLlApi.Wire - Received UDP packet on 14.50.245.9:5065 ,source 14.50.245.20:5065 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 14.50.245.9:5065;branch=z9hG4bKM3X51yKL9BEW5v0Kudc5Dw~~128,SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F To: <sip:2003@14.50.245.9>;tag=93896~37db7c49-96d4-4c4c-a223-626b2c74c16a-16919968 Date: Thu, 17 Aug 2017 17:48:52 GMT Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence Server: Cisco-CUCM11.5 Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP Supported: X-cisco-srtp-fallback Supported: Geolocation Session-ID: 1e6e772300105000a00084b517ae1a83;remote=c07cdfa83b8f7c373757cf842ab93896 P-Asserted-Identity: "Alerting JM1 - 2003" <sip:2003@14.50.245.20> Remote-Party-ID: "Alerting JM1 - 2003" <sip:2003@14.50.245.20>;party=called;screen=yes;privacy=off Contact: <sip:2003@14.50.245.20:5065>;+u.sip!devicename.ccm.cisco.com="SEP84B517AE1A83" Content-Length: 0 --- end of packet --- [CT_CALLBACK.15] DEBUG 2017.08.17 13:48:52:285 DsSipLlApi.Wire - Sending UDP packet on 14.50.245.9:32789, destination 14.50.245.6:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 To: <sip:2003@14.50.245.9>;tag=93896~37db7c49-96d4-4c4c-a223-626b2c74c16a-16919968 From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F Contact: <sip:2003@14.50.245.20:5065>;+u.sip!devicename.ccm.cisco.com="SEP84B517AE1A83" Remote-Party-ID: "Alerting JM1 - 2003" <sip:2003@14.50.245.20>;party=called;screen=yes;privacy=off Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Content-Length: 0 Date: Thu, 17 Aug 2017 17:48:52 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence Server: Cisco-CUCM11.5 Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP Supported: X-cisco-srtp-fallback Supported: Geolocation Session-ID: 1e6e772300105000a00084b517ae1a83;remote=c07cdfa83b8f7c373757cf842ab93896 P-Asserted-Identity: "Alerting JM1 - 2003" <sip:2003@14.50.245.20> [DsTransportListener-3] DEBUG 2017.08.17 13:48:54:292 DsSipLlApi.Wire - Received UDP packet on 14.50.245.9:5065 ,source 14.50.245.20:5065 SIP/2.0 200 OK Via: SIP/2.0/UDP 14.50.245.9:5065;branch=z9hG4bKM3X51yKL9BEW5v0Kudc5Dw~~128,SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F To: <sip:2003@14.50.245.9>;tag=93896~37db7c49-96d4-4c4c-a223-626b2c74c16a-16919968 Date: Thu, 17 Aug 2017 17:48:52 GMT Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence, kpml Supported: replaces Server: Cisco-CUCM11.5 Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP Supported: X-cisco-srtp-fallback Supported: Geolocation Session-Expires: 1800;refresher=uas Require: timer Session-ID: 1e6e772300105000a00084b517ae1a83;remote=c07cdfa83b8f7c373757cf842ab93896 P-Asserted-Identity: "CLID JM1 - 2003" <sip:2003@14.50.245.20> Remote-Party-ID: "CLID JM1 - 2003" <sip:2003@14.50.245.20>;party=called;screen=yes;privacy=off Contact: <sip:2003@14.50.245.20:5065>;+u.sip!devicename.ccm.cisco.com="SEP84B517AE1A83" Content-Type: application/sdp Content-Length: 258 v=0 o=CiscoSystemsCCM-SIP 93896 1 IN IP4 14.50.245.20 s=SIP Call c=IN IP4 14.50.245.254 b=TIAS:8000 b=AS:8 t=0 0 m=audio 16502 RTP/AVP 18 101 a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- end of packet --- [CT_CALLBACK.15] DEBUG 2017.08.17 13:48:54:293 DsSipLlApi.Wire - Sending UDP packet on 14.50.245.9:32789, destination 14.50.245.6:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 14.50.245.6:5060;branch=z9hG4bK2A5763 To: <sip:2003@14.50.245.9>;tag=93896~37db7c49-96d4-4c4c-a223-626b2c74c16a-16919968 From: <sip:1001@14.50.245.6>;tag=4E329FEC-A9F Contact: <sip:2003@14.50.245.20:5065>;+u.sip!devicename.ccm.cisco.com="SEP84B517AE1A83" Require: timer Remote-Party-ID: "CLID JM1 - 2003" <sip:2003@14.50.245.20>;party=called;screen=yes;privacy=off Call-ID: 2A7BE22B-82AB11E7-83AEAE0B-F940DC75@14.50.245.6 CSeq: 101 INVITE Content-Length: 258 Date: Thu, 17 Aug 2017 17:48:52 GMT Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence, kpml Supported: replaces Supported: X-cisco-srtp-fallback Supported: Geolocation Server: Cisco-CUCM11.5 Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP Session-Expires: 1800;refresher=uas Session-ID: 1e6e772300105000a00084b517ae1a83;remote=c07cdfa83b8f7c373757cf842ab93896 P-Asserted-Identity: "CLID JM1 - 2003" <sip:2003@14.50.245.20> Content-Type: application/sdp v=0 o=CiscoSystemsCCM-SIP 93896 1 IN IP4 14.50.245.20 s=SIP Call c=IN IP4 14.50.245.254 b=TIAS:8000 b=AS:8 t=0 0 m=audio 16502 RTP/AVP 18 101 a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
SIP-Wire-Log shows the normal SIP messaging up to the 200 Okay for both call legs.