SIP flashcards

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SIP Application signaling for sessions. The base specification is RFC 3261. What SIP is for
SDP Describes media. Offer/answer is RFC 3264. What SIP is for
RTP Carries the media packets after the answer names the port. What SIP is for
100 Trying Hop-by-hop provisional. A stateful proxy generates it and does not relay one. What SIP is for
5060 Default port for SIP over UDP and SIP over TCP. Ports, transports, and where a request can go
5061 Default port for SIP over TLS. Ports, transports, and where a request can go
Content-Length On a stream transport, this is how the parser finds the end of the message. Ports, transports, and where a request can go
WS / WSS WebSocket transport tokens from RFC 7118. The SIP methods do not change. Ports, transports, and where a request can go
tel:+15551212 A global telephone URI. The scheme is RFC 3966. E.164, sip URIs, and tel URIs
user=phone The sip user part should be read as a phone number. E.164, sip URIs, and tel URIs
E.164 The ITU-T numbering plan. The URI citation in this course is RFC 3966. E.164, sip URIs, and tel URIs
Request-URI Where this hop sends the request. Retargeting changes it and leaves To in place. E.164, sip URIs, and tel URIs
Standards Track The status of RFC 3261. It is the base specification. How to read an RFC
Section 20 Header field catalog, including the compact names defined there. How to read an RFC
Section 21 Response code catalog. How to read an RFC
Section 25 ABNF. Use it when a message looks almost right. How to read an RFC
RFC 5411 Informational hitchhiker's guide. A map, not the base spec. The hitchhiker guide is a map
RFC 3262 PRACK and reliable provisional responses. The hitchhiker guide is a map
RFC 4028 Session timers. Not a substitute chapter inside 5411. The hitchhiker guide is a map
RFC 6026 Corrects INVITE 2xx transaction handling after 3261. The hitchhiker guide is a map
IETF Trust Legal provisions for IETF documents. Read them before copying protocol text. Licenses, codecs, and what you may copy
Opus RFC 6716, plus the Opus license page and the IETF IPR disclosures. Licenses, codecs, and what you may copy
G.711 ITU-T recommendation for PCMU and PCMA. Not an IETF RFC. Licenses, codecs, and what you may copy
G.729 ITU-T recommendation. Verify patent status. Do not state an expiry date. Licenses, codecs, and what you may copy
z9hG4bK Magic cookie at the start of a branch parameter. Reading the first INVITE
Max-Forwards: 70 The value a UAC SHOULD put on a request it originates. Reading the first INVITE
No To tag An initial request is outside a dialog, so To has no tag. Reading the first INVITE
RFC 5737 Documentation IPv4 ranges used for the examples in this course. Reading the first INVITE
UAC Creates a request. Defined with the rest of the user-agent rules. Roles a trace is actually showing
Stateful proxy Transaction state, own Via branch, own 100 Trying. Roles a trace is actually showing
Redirect 3xx and a Contact, then it is out of the call. Roles a trace is actually showing
B2BUA UAS plus UAC. Expect a second dialog. See RFC 7092. Roles a trace is actually showing
INVITE / ACK / BYE Set up, complete the handshake, and end a confirmed dialog. Methods and status codes
CANCEL Abandon an unanswered INVITE. It is not BYE. Methods and status codes
401 vs 407 UAS or registrar challenge versus proxy challenge. Methods and status codes
1xx Provisional. The transaction is still waiting for a final response. Methods and status codes
v / f / t / i Via, From, To, Call-ID. Headers and compact forms
m / l / c / k Contact, Content-Length, Content-Type, Supported. Headers and compact forms
o and u Event and Allow-Events. RFC 6665. Headers and compact forms
r and b Refer-To and Referred-By. RFC 3515. Headers and compact forms
Branch Transaction identifier. Cookie z9hG4bK plus a unique token. Transactions and the branch cookie
Retransmission Same message, same branch, sent again by the transaction timer. Transactions and the branch cookie
2xx ACK New transaction, new branch, end to end. CSeq number matches the INVITE. Transactions and the branch cookie
Timer B INVITE transaction timeout, 64 times T1. No ACK is generated. Transactions and the branch cookie
Call-ID + tags The dialog id. Both tags are required once the dialog exists. Dialog identity
Early Provisional response with a To tag. Not confirmed yet. Dialog identity
Confirmed A 2xx to INVITE. BYE is now the teardown. Dialog identity
CSeq Increments for new in-dialog requests. ACK and CANCEL keep the INVITE number. Dialog identity
T1 = 500 ms Default RTT estimate. Timer A starts here on UDP. T1, the transaction timeout, and Timer C
64*T1 Timer B for INVITE and Timer F for non-INVITE. T1, the transaction timeout, and Timer C
T2 = 4 s, T4 = 5 s Retransmit cap, and how long a message may remain in the network. T1, the transaction timeout, and Timer C
Timer C Proxy timer for an unanswered INVITE. More than 3 minutes. T1, the transaction timeout, and Timer C
Via transport UDP, TCP, TLS, WS, or WSS for this hop. UDP, TCP, TLS, and WebSocket
received / rport Response address and, with RFC 3581, the source port. UDP, TCP, TLS, and WebSocket
session protocol sipv2 Dial-peer command that selects SIP for that peer. UDP, TCP, TLS, and WebSocket
ip address trusted list Sources allowed to present SIP. See the toll-fraud guidance. UDP, TCP, TLS, and WebSocket
Non-2xx ACK Same branch as that hop's INVITE. Generated by the client transaction. Two different ACK rules
2xx ACK New branch, dialog route set, end to end. Two different ACK rules
CSeq: 1 ACK Same number as the INVITE. The method token changes to ACK. Two different ACK rules
603 then ACK Forward the 603. ACK it on this hop. Do not relay the ACK. Two different ACK rules
Record-Route Inserted by a proxy that must see later requests in the dialog. Record-Route and the route set
lr Required on that URI. It selects loose routing. Record-Route and the route set
UAC reverses Response Record-Route, reversed, becomes the caller's route set. Record-Route and the route set
Contact The remote target. It becomes the mid-dialog Request-URI. Record-Route and the route set
Supported: 100rel The UAC can do reliable provisionals. PRACK and reliable provisionals
RSeq Identifies the reliable provisional. PRACK and reliable provisionals
RAck RSeq, then the CSeq number and method being acknowledged. PRACK and reliable provisionals
PRACK A new transaction. Its success response is 200 OK, not ACK. PRACK and reliable provisionals
UPDATE Session refresh or change. Own transaction. RFC 3311. UPDATE before the INVITE completes
No ACK 200 OK finishes UPDATE. ACK is reserved for INVITE. UPDATE before the INVITE completes
Early dialog UPDATE can run before the first INVITE completes. UPDATE before the INVITE completes
RFC 6141 Read this before changing the remote target. UPDATE before the INVITE completes
Session-Expires Seconds until the session must be refreshed, plus an optional refresher. Session-Expires and 422
Min-SE Minimum interval. Never below 90 seconds. Session-Expires and 422
422 Session Interval Too Small. Retry with the Min-SE from the response. Session-Expires and 422
refresher uac or uas. That side sends re-INVITE or UPDATE. Session-Expires and 422
CANCEL Same branch as the unanswered INVITE. CSeq number unchanged. CANCEL, BYE, and re-INVITE
487 Final response to the INVITE that was cancelled. Then ACK. CANCEL, BYE, and re-INVITE
BYE Ends a confirmed dialog. Higher CSeq. Both tags. CANCEL, BYE, and re-INVITE
re-INVITE Mid-dialog INVITE. See RFC 6141 if Contact changes the target. CANCEL, BYE, and re-INVITE
Refer-To The request the recipient should send. Compact form r. REFER and Replaces
202 Accepted REFER was accepted. Watch NOTIFY for the real result. REFER and Replaces
Replaces call-id, to-tag, from-tag of the dialog to take over. REFER and Replaces
sipfrag A status line in the NOTIFY body, such as SIP/2.0 200 OK. REFER and Replaces
To on REGISTER The address-of-record being bound. REGISTER and bindings
Contact The binding: where requests for that address-of-record can go. REGISTER and bindings
423 Interval Too Brief. Read Min-Expires and retry. REGISTER and bindings
Contact: * With Expires: 0, removes all bindings for the address-of-record. REGISTER and bindings
NAPTR Selects the SIP service, such as SIP+D2U or SIP+D2T. Locating a server with RFC 3263
SRV Host and port for the selected service. Locating a server with RFC 3263
A / AAAA Addresses for the SRV target. Locating a server with RFC 3263
Numeric IP No domain lookup. Send to that address. Locating a server with RFC 3263
lr Loose routing. Pop Route, keep the Request-URI. Loose routing and Max-Forwards
Max-Forwards Starts at 70. Decremented by each proxy. Loose routing and Max-Forwards
483 Too Many Hops. The request arrived with Max-Forwards 0. Loose routing and Max-Forwards
482 Loop Detected from a Via loop, not from the hop budget. Loose routing and Max-Forwards
reg-id Identifies one flow. A new value is a new flow. Outbound, reg-id, and keepalives
+sip.instance URN for the UA instance, stable across flows. Outbound, reg-id, and keepalives
CRLF keepalive Double CRLF on connection-oriented transports. Single CRLF answers. Outbound, reg-id, and keepalives
STUN keepalive The UDP keepalive in the outbound specification. Outbound, reg-id, and keepalives
Path Inbound vector stored with the binding. RFC 3327. Path and Service-Route
Service-Route Outbound preloaded route from the registrar. RFC 3608. Path and Service-Route
lr on Path The edge URI is loose-routed, same as Record-Route. Path and Service-Route
Preloaded Route The UA puts Service-Route into Route on a new INVITE. Path and Service-Route
pub-gruu Stable URI for one UA instance. GRUU and a single instance
temp-gruu Instance URI that hides the address-of-record. GRUU and a single instance
+sip.instance The URN the registrar binds to those GRUUs. GRUU and a single instance
gr URI parameter that marks a GRUU. GRUU and a single instance
Fork One incoming INVITE, one client transaction per target. Forking and early dialogs
Early dialog A To tag on a provisional response. One per branch. Forking and early dialogs
Multiple 200 OK Forward each. The UAC ACKs each and may BYE the extras. Forking and early dialogs
6xx Cancel the other pending branches. Do not start new ones. Forking and early dialogs
RFC 8866 Current SDP. It obsoletes RFC 4566. Trust the RFC Editor status banner. The lines in an SDP body
c= Connection address where media should be sent. The lines in an SDP body
m= One media stream: type, port, protocol, and payload types. The lines in an SDP body
a= An attribute. After m=, it belongs to that stream. The lines in an SDP body
Same count The answer has exactly as many m-lines as the offer. Same m-lines, same order
Same order Line n in the answer answers line n in the offer. Same m-lines, same order
Port 0 Rejects that stream and keeps its position. Same m-lines, same order
RFC 6337 Usage examples and the pitfalls of rebuilding SDP. Same m-lines, same order
Encoding name Match opus to opus even when the payload type numbers differ. rtpmap names and fmtp parameters
0 / 8 / 9 / 18 PCMU/8000, PCMA/8000, G722/8000, G729/8000. rtpmap names and fmtp parameters
G722/8000 RTP clock is 8000. Audio sample rate is 16 kHz. rtpmap names and fmtp parameters
fmtp Parameters for one payload type. Do not guess unknown ones. rtpmap names and fmtp parameters
sendrecv Send and receive. The usual active call. Direction attributes and hold
sendonly Send, and do not receive. A common hold. Direction attributes and hold
recvonly Receive only. The usual answer to sendonly. Direction attributes and hold
inactive Neither direction. Also a common hold. Direction attributes and hold
m=audio 0 Rejects that audio stream. Keep proto and a format. Port 0 rejects a stream
Same index A rejected line is still answered in its original position. Port 0 rejects a stream
No common codec Port 0 on that m-line, not a deleted body. Port 0 rejects a stream
Offer port 0 The answer port is also 0. Port 0 rejects a stream
183 + SDP Usual early-media pattern. Not true of every 183. Early media and 183
180 Ringing. Local ringback when this message has no SDP. Early media and 183
100rel Makes a provisional response reliable. PRACK it. Early media and 183
Two bodies Compare 183 SDP with the 200 OK SDP before you lock the port. Early media and 183
RFC 6337 Offer/answer usage and the mistakes to stop repeating. Offer/answer pitfalls in RFC 6337
sess-version Increase it when the SDP changes. Offer/answer pitfalls in RFC 6337
Delayed offer No SDP in the INVITE. Offer in the reliable response. Answer in the ACK. Offer/answer pitfalls in RFC 6337
Do not rebuild Edit m-lines in place. Do not drop, swap, or renumber them. Offer/answer pitfalls in RFC 6337
V=2 RTP version is 2. The RTP header
Sequence Plus one for every packet from that source. The RTP header
Timestamp Media clock. 20 ms of G.711 or G.722 steps by 160. The RTP header
SSRC 32-bit source id chosen by the sender. The RTP header
PT 0 PCMU/8000, static. Static and dynamic payload types
PT 8 PCMA/8000, static. Not the same as PCMU. Static and dynamic payload types
PT 9 G722/8000. Sample rate is still 16 kHz. Static and dynamic payload types
PT 18 G729/8000, static. Static and dynamic payload types
G.711 PCMU/8000 and PCMA/8000. Read ITU-T G.711. They are not interchangeable. G.711, G.729, and Opus
G.729 rtpmap G729/8000, static PT 18. Read ITU-T G.729. This course states no licensing result. G.711, G.729, and Opus
Opus RFC 6716. Read the RFC and the license. G.711, G.729, and Opus
IPR Check IETF IPR disclosures. This course states no legal result. G.711, G.729, and Opus
SR Sender report. Used when this participant has sent RTP. RTCP sender and receiver reports
RR Receiver report. Fraction lost, cumulative loss, jitter. RTCP sender and receiver reports
Port pair RTP on an even port, RTCP on the next higher port, by default. RTCP sender and receiver reports
Not SIP 200 OK does not carry fraction lost. RTCP sender and receiver reports
RFC 4733 telephone-event. Obsoletes RFC 2833. Trust the status banner. telephone-event and other digit paths
Events 0-15 DTMF. Event 16 is flash. telephone-event and other digit paths
Dynamic PT Match the name telephone-event in SDP. telephone-event and other digit paths
Not INFO RFC 6086 is a method. application/dtmf-relay is a vendor body. telephone-event and other digit paths
a=ptime:20 20 ms of media per packet. SDP, not RTP. ptime, packet rate, and EF
50 and 160 G.711 at 20 ms: 50 packets/s, 160 payload bytes, plus headers. ptime, packet rate, and EF
EF Voice service class in RFC 4594. ptime, packet rate, and EF
RFC 3246 Defines the EF per-hop behavior. No router config in this lesson. ptime, packet rate, and EF
200 OK Offer/answer finished. It does not prove both RTP directions. One-way audio
sendonly answer Answerer sends, does not receive. One-way by negotiation. One-way audio
c= and port The destination you must capture. One-way audio
SR from one side The other side is not sending, according to RTCP. One-way audio
401 From the UAS. Challenge is WWW-Authenticate. Retry with Authorization. 401, 407, and digest
407 From the proxy. Challenge is Proxy-Authenticate. Retry with Proxy-Authorization. 401, 407, and digest
RFC 7616 HTTP digest algorithms. Read the algorithm parameter. 401, 407, and digest
Stop here Identify the header. Do not compute a digest in this lesson. 401, 407, and digest
RFC 3261 §18 SIP over TLS. Same methods. Via transport TLS. TLS and domain certificates
5061 Usual TLS port in RFC 3261. TLS and domain certificates
RFC 5922 The certificate must identify the SIP domain. TLS and domain certificates
Not SRTP TLS on SIP does not select the media keying method. TLS and domain certificates
RFC 3711 SRTP. Protects RTP. Profile RTP/SAVP. SRTP and two keying methods
RFC 4568 SDES. a=crypto in SDP. Protect the signaling. SRTP and two keying methods
RFC 5764 DTLS-SRTP. Keys from DTLS. Fingerprint in SDP. SRTP and two keying methods
Per leg Do not assume two peers share one keying method. SRTP and two keying methods
RFC 3323 Privacy header. id withholds a network-asserted identity. Privacy and asserted identity
PPI P-Preferred-Identity. A hint from the UA inside the trust domain. Privacy and asserted identity
PAI P-Asserted-Identity. Trusted networks only. Not a signature. Privacy and asserted identity
Strip Remove PAI and PPI when the request leaves the trust domain. Privacy and asserted identity
RFC 8224 STIR. Identity header in SIP. STIR, PASSporT, and SHAKEN
RFC 8225 PASSporT. STIR, PASSporT, and SHAKEN
RFC 8226 Certificates for STIR. STIR, PASSporT, and SHAKEN
Levels A/B/C ATIS SHAKEN. Confirm the current ATIS document. See also the FCC page. STIR, PASSporT, and SHAKEN
Scanning Unsolicited SIP from addresses you did not provision. Close the exposure. Scanning, hijack, and toll fraud
Hijack Unexpected Contact on an address-of-record. Authenticate REGISTER. Scanning, hijack, and toll fraud
Toll fraud Unauthorized calls that run up charges. Authorize the source. Scanning, hijack, and toll fraud
FCC page Call authentication is not a substitute for outbound authorization. Scanning, hijack, and toll fraud
Allow list Named peers only. See the CME toll-fraud guide for trusted sources. Hardening a SIP edge
Digest 401 uses Authorization. 407 uses Proxy-Authorization. Hardening a SIP edge
RFC 5922 Certificate name equals the SIP domain. Hardening a SIP edge
Per-leg SRTP Do not copy keying from one peer to another. Hardening a SIP edge
Embedded Via, Contact, and SDP c= carry addresses NAT does not rewrite. Embedded addresses and NAT
c= Media destination. A private value is non-routable on the public Internet. Embedded addresses and NAT
200 OK Signaling can succeed while RTP still cannot go where c= points. Embedded addresses and NAT
RFC 5737 203.0.113.10 and the other lab addresses are documentation ranges. Embedded addresses and NAT
RFC 3581 rport. Symmetric SIP response routing. rport and symmetric responses
received Source address the server saw. rport and symmetric responses
rport value Source port the server saw. rport and symmetric responses
Not SDP rport does not rewrite c=. rport and symmetric responses
RFC 8489 Current STUN. Obsoletes RFC 5389. Trust the status banner. STUN server-reflexive addresses
Binding Request that asks what address the server sees. STUN server-reflexive addresses
XOR-MAPPED-ADDRESS The reflexive address in the success response. STUN server-reflexive addresses
Not SIP A STUN server does not store a SIP Contact. STUN server-reflexive addresses
RFC 8656 Current TURN. Obsoletes RFC 5766. Trust the status banner. TURN relays
Allocate Reserves a relayed transport address. TURN relays
Relay Peers send media to the relay. TURN forwards it. TURN relays
Not SIP Allocate is not REGISTER and not an SDP rewrite. TURN relays
RFC 8445 Current ICE. Obsoletes RFC 5245. Trust the status banner. ICE candidate types
host Local interface. ICE candidate types
srflx / relay STUN reflexive address, and TURN relay. ICE candidate types
prflx Discovered during a connectivity check. ICE candidate types
ALG In-path SIP and SDP rewrite. Often corrupts the call. The SIP ALG problem
Length If Content-Length does not match the body, the edit is corrupt. The SIP ALG problem
TLS The ALG cannot rewrite what it cannot see. The SIP ALG problem
Disable it Turn the ALG off. Do not stack a second hidden editor. The SIP ALG problem
Latch Learn the remote RTP address from the first packet instead of trusting c=. Media latching on an SBC
RFC 5853 SBC requirements. Not an IETF algorithm for latching. Media latching on an SBC
Authenticated Latch only after signaling is accepted, and only toward the expected side. Media latching on an SBC
ICE RFC 8445 is the specified candidate check when both ends use it. Media latching on an SBC
RFC 5853 Informational requirements from SBC deployments. Not an SBC protocol. RFC 5853 is requirements, not a protocol
No SBC method INVITE and OPTIONS stay as RFC 3261 defined them. RFC 5853 is requirements, not a protocol
Datasheet Session and transcode limits are platform-specific. RFC 5853 is requirements, not a protocol
B2BUA The usual SIP role behind an SBC. Taxonomy is RFC 7092. RFC 5853 is requirements, not a protocol
B2BUA UAS plus UAC. Two dialogs for one call. An SBC is commonly a B2BUA
Pure proxy Forwards the request and adds Via. An SBC is commonly a B2BUA
Call-ID Often different on the two legs of an SBC. An SBC is commonly a B2BUA
RFC 3261 Defines the B2BUA as UAS concatenated with UAC. An SBC is commonly a B2BUA
Via Identifies the previous SIP hop. Hide the internal one. Topology hiding rewrites the addresses you leak
Contact Dialog target. An SBC usually substitutes its own. Topology hiding rewrites the addresses you leak
c= SDP connection address. Anchoring rewrites it. Topology hiding rewrites the addresses you leak
User-Agent A common topology leak. Stripping it is a product rule. Topology hiding rewrites the addresses you leak
Anchor RTP terminates on the SBC. SDP c= names the SBC. Anchor, bypass, transcode, and DTMF
Bypass SIP through the SBC. Media between endpoints. Anchor, bypass, transcode, and DTMF
Transcode Decode and encode. Capacity is on the datasheet. Anchor, bypass, transcode, and DTMF
telephone-event RFC 4733 DTMF in RTP. Anchor, bypass, transcode, and DTMF
OPTIONS Capability and liveness request. RFC 3261. OPTIONS keepalives
CSeq method Must match the start line. OPTIONS stays OPTIONS. OPTIONS keepalives
200 OK The peer answered. The call is a later transaction. OPTIONS keepalives
Interval Configured on the product. Not an RFC constant. OPTIONS keepalives
Admission Refuse new work when the configured budget is exhausted. Call admission stays on the datasheet
503 Service Unavailable. Confirm the product’s actual code. Call admission stays on the datasheet
Datasheet The only source for this platform’s session limit. Call admission stays on the datasheet
No CPS formula Do not invent calls-per-second math. Call admission stays on the datasheet
Active The unit currently processing the dialogs. Active and standby, without a made-up timer
Standby Ready to take the role. Not a SIP method. Active and standby, without a made-up timer
No timer here Do not invent a failover interval. Active and standby, without a made-up timer
Dialog state A B2BUA holds it. Survival is a product question. Active and standby, without a made-up timer
CUBE Cisco Unified Border Element. An IOS XE B2BUA. CUBE is an IOS XE B2BUA
IOS XE The operating system whose voice stack CUBE uses. CUBE is an IOS XE B2BUA
SIP profiles Header modifications. Guide for 17.6 and later. CUBE is an IOS XE B2BUA
B2BUA The role RFC 7092 taxonomizes. CUBE is an IOS XE B2BUA
dial-peer voice N voip Creates the VoIP dial peer numbered N. Dial peers select the SIP next hop
session protocol sipv2 This dial peer speaks SIP. Dial peers select the SIP next hop
session target ipv4: The next-hop IPv4 address. Dial peers select the SIP next hop
192.0.2.40 Carrier address in this synthetic. Dial peers select the SIP next hop
sip-profiles voice class sip-profiles N holds the rules. voice class sip-profiles
remove request INVITE sip-header Diversion remove. voice class sip-profiles
After 15.4 Inbound profile support, per the use-case technote. voice class sip-profiles
CSA tester Try the profile before loading it. voice class sip-profiles
Delayed offer INVITE without SDP. Offer comes later. Early offer and delayed offer
Early offer The INVITE already contains the offer. Early offer and delayed offer
early-offer forced Dial-peer command that forces an outbound offer. Early offer and delayed offer
ACK answer Used when the offer arrived in the 200 OK. Early offer and delayed offer
SIP trunk CUCM’s adjacency to CUBE or another peer. CUCM trunks, SIP profiles, and Lua
CUCM SIP profile Not voice class sip-profiles. CUCM trunks, SIP profiles, and Lua
Lua Normalization scripts exist. Open the admin guide. CUCM trunks, SIP profiles, and Lua
MTP required Confirm the guide for that version before checking it. CUCM trunks, SIP profiles, and Lua
SIP integration The documented Unity Connection path. Open your version’s guide. Unity Connection SIP integration
Diversion RFC 5806 redirecting number. Reason unconditional in the lab. Unity Connection SIP integration
History-Info RFC 7044. Use it if the guide asks for it. Unity Connection SIP integration
Not SCCP-by-guess Do not invent SCCP as the current default. Unity Connection SIP integration
debug ccsip messages Shows sent and received SIP. debug ccsip messages
sip-profiles debug debug ccsip feature sip-profiles. debug ccsip messages
CSA SIP Profile Tester in Collaboration Solutions Analyzer. debug ccsip messages
Sent / Received Direction markers around each SIP message. debug ccsip messages
realm Routing and policy context on the Oracle SBC. Oracle SBC objects
session-agent Named next hop. Oracle SBC objects
in-manipulationid Which sip-manipulation runs on arrival. Oracle SBC objects
out-manipulationid Which sip-manipulation runs on the way out. Oracle SBC objects
header-rule One rule inside a sip-manipulation. Header rules run in order
delete Removes the named header. Lab rule strip-server. Header rules run in order
store Keeps a value for a later rule. Run it first. Header rules run in order
reject An HMR action. Read the release guide for the status. Header rules run in order
SWe Edge Edge product. Family includes SBC 1000 and SBC 2000. Ribbon Edge, Core, and public validation lists
SBC Core A separate family. Ribbon Edge, Core, and public validation lists
ACS snapshot 5110, 5210, 5400, 7000, and SWe for that validation. Ribbon Edge, Core, and public validation lists
No invented CLI Admin guides live on the documentation portal. Ribbon Edge, Core, and public validation lists
Action Subject Which header or element the rule touches. AudioCodes manipulation in software 7.6
Action Type Add, Remove, Modify, prefix, suffix, or Normalize. AudioCodes manipulation in software 7.6
IP group Inbound or outbound assignment of the set. AudioCodes manipulation in software 7.6
Release menu The clicks change. Use the 7.6 manual you are running. AudioCodes manipulation in software 7.6
CUBE line request INVITE sip-header NAME remove. Three manipulation grammars
Oracle block header-rule, header-name, action delete. Three manipulation grammars
AudioCodes row Action Type Remove on an IP group. Three manipulation grammars
Do not paste across Each grammar is parsed by its own product. Three manipulation grammars
PAI P-Asserted-Identity, RFC 3325. Identity at a trust boundary
Privacy RFC 3323. id asks the network to withhold identity. Identity at a trust boundary
Diversion RFC 5806 redirecting header. Identity at a trust boundary
History-Info RFC 7044. Map it only as the interop guide says. Identity at a trust boundary
RFC 5922 SIP domain certificates. Certificate names and Contact
Teams Contact FQDN. An IP draws 403. Certificate names and Contact
Teams wildcard CN or SAN wildcards are supported for Direct Routing. Certificate names and Contact
Webex LGW wildcard Not supported for Local Gateway. Certificate names and Contact
pstnhub sip, sip2, and sip3.pstnhub.microsoft.com. Teams Direct Routing signaling
5061 Direct Routing signaling port. Teams Direct Routing signaling
Verified domain Not a name ending in onmicrosoft.com. Teams Direct Routing signaling
No RFC 5923 The SIP proxy does not support connection aliasing. Teams Direct Routing signaling
OPTIONS 200 Marks the Direct Routing SBC healthy. OPTIONS health and SRTP media
CSeq Method token matches the request. OPTIONS, not INVITE. OPTIONS health and SRTP media
Table cell Checked 2026-10-14: CSeq: 1 INVITE. Do not copy it. OPTIONS health and SRTP media
SRTP Media processors use SRTP. RFC 3711. OPTIONS health and SRTP media
SIP TLS Signaling from an IOS XE Local Gateway to Webex. Webex Calling Local Gateway
SRTP Media for that IOS XE Local Gateway. Webex Calling Local Gateway
Certificate trunk How a third-party SBC is supported. Webex Calling Local Gateway
ByoPSTN Bring your own PSTN. Not Cloud Connected PSTN. Webex Calling Local Gateway
Cisco Calling Plan One of the three organization-article choices. Three PSTN choices, one per location
Cloud Connected PSTN A cloud partner through Webex. Not the Local Gateway. Three PSTN choices, one per location
Local Gateway Premises-based PSTN. One connection on the location. Three PSTN choices, one per location
ByoPSTN Bring your own PSTN. Three PSTN choices, one per location
Managed PSTN Ordinary Connect calls. No customer SIP trunk required. Amazon Connect voice boundaries
SIPREC External voice into the analytics connector. Amazon Connect voice boundaries
Voice Connector Chime SDK SIP trunking service. Amazon Connect voice boundaries
Not a bridge Carrier through Voice Connector into Connect needs specialist approval. Amazon Connect voice boundaries
BYO SIP Your ITSP or gateway connects to Five9 SBCs. Five9 BYO SIP and Twilio trunks
Entitled groups Self-service exists for entitled BYO SIP trunk groups. Five9 BYO SIP and Twilio trunks
Termination URI Must appear in the Request-URI or Twilio drops the request. Five9 BYO SIP and Twilio trunks
sip not sips Origination URLs use sip. sips is not supported. Five9 BYO SIP and Twilio trunks
BYOC Cloud Public internet SIP. RTP crosses the internet. Genesys BYOC and NICE CXone SIP
BYOC Premises Customer-managed Edge. Hybrid media is allowed. Genesys BYOC and NICE CXone SIP
December 1, 2026 Genesys Hardware Solution only. Genesys BYOC and NICE CXone SIP
CXone ports 5060 and 5061, plus dynamic RTP/SRTP. Genesys BYOC and NICE CXone SIP
Call-ID Same value for the dialog. A B2BUA may use a new one on the other leg. Match a ladder before you change anything
CSeq Integer plus method. The ACK for an INVITE keeps the INVITE's number. Match a ladder before you change anything
Via branch Transaction hop id. Look for the z9hG4bK cookie. Match a ladder before you change anything
2xx ACK End to end. New branch. RFC 6026 corrects the transaction rules around it. Match a ladder before you change anything
180 Ringing. Provisional alerting. Status codes you will actually see
183 Session Progress. Provisional, often where early media shows up. Status codes you will actually see
487 Request Terminated. The INVITE's answer after CANCEL. Status codes you will actually see
606 Not Acceptable. Global. 488 is only this UA. Status codes you will actually see
sip Display filter for SIP packets. Field names live on the SIP reference page. Display filters for SIP and RTP
rtp Display filter for RTP packets. Field names live on the RTP reference page. Display filters for SIP and RTP
sip.Method Safe display field for the SIP method. Display filters for SIP and RTP
Confirm CSeq sip.CSeq.method is common. Confirm it on the SIP display-filter reference. Display filters for SIP and RTP
ccsip messages IOS voice debug that prints SIP send and receive. IOS ccsip and ccapi debug
ccapi inout IOS voice debug of the call-control API. Not a SIP RFC. IOS ccsip and ccapi debug
Sent / Received Direction on the debug. The SIP start line comes after it. IOS ccsip and ccapi debug
Synthetic Lab output in this track is synthetic. Customer logs stay out. IOS ccsip and ccapi debug
488 Not Acceptable Here. This UA. Often codec or SDP. 488 codecs and 422 session timers
422 Session Interval Too Small. RFC 4028. Read Min-SE. 488 codecs and 422 session timers
telephone-event RFC 4733. Offer it on purpose if you need DTMF. 488 codecs and 422 session timers
Offer/answer The answer is a subset. No shared audio codec tends to 488. 488 codecs and 422 session timers
Name mismatch Handshake fails. No SIP status from the peer. TLS names, trust anchors, and version
Untrusted CA Handshake fails. Fix the trust anchor. TLS names, trust anchors, and version
TLS 1.2+ Teams Direct Routing floor in the OPTIONS troubleshooting page. TLS names, trust anchors, and version
FQDN Contact Required toward Teams. An address literal draws 403. TLS names, trust anchors, and version
Category Read it on the header. 3261 is Standards Track. 6881 is BCP. Read an RFC header before you cite it
Obsoletes The new RFC names the one it replaces. 8866 and 4566 are the SDP pair here. Read an RFC header before you cite it
Updates A correction that leaves the base in force. Check 6026's header. Read an RFC header before you cite it
rfc-editor.org The header you cite. Months in this lesson are not invented. Read an RFC header before you cite it
Trusted list Who may present VoIP signaling. CME and SRST each have a page. The toll-fraud class, stated defensively
401 / 407 UAS and proxy digest challenges. Secrets stay unprinted. The toll-fraud class, stated defensively
Numbering plan Dial-peers match the plan, including whether international destinations are in it. The toll-fraud class, stated defensively
STIR context FCC call authentication and RFC 8224. A different question from the source list. The toll-fraud class, stated defensively
trusted list Addresses allowed to present inbound VoIP. CME page or SRST page. Trusted addresses and source restrictions
authenticate The check that uses the list. Trusted addresses and source restrictions
Two pages CME and SRST differ. Quote no shared list size. Trusted addresses and source restrictions
Border pair 198.51.100.30 and 192.0.2.40 in the teaching snippet. Trusted addresses and source restrictions
urn:service:sos RFC 5031 service URN for emergency calling. Emergency calling, location, and the FCC page
RFC 6881 BCP for emergency calling services. Emergency calling, location, and the FCC page
Geolocation RFC 6442 location conveyance. Emergency calling, location, and the FCC page
fcc-mlts Kari's Law and RAY BAUM's Act. The page wins over a retelling. Emergency calling, location, and the FCC page
7866 Session recording protocol. SRC forks to an SRS. Disclosed recording with SIPREC
7865 Recording metadata. Read it for fields. Disclosed recording with SIPREC
Disclosed Quality monitoring and compliance, with the parties on notice. Disclosed recording with SIPREC
Retention A legal question. Outside the RFC. Disclosed recording with SIPREC
EF Expedited Forwarding per-hop behavior. RFC 3246. Voice, EF, and what the PHB does not design
Telephony The service class RFC 4594 maps to EF. Voice, EF, and what the PHB does not design
Codepoint Use the value RFC 3246 recommends. Do not invent a second one. Voice, EF, and what the PHB does not design
Not a LAN Queues and campus policy stay in the approved design. Voice, EF, and what the PHB does not design
OPTIONS Liveness. Alarm when the trunk stops answering. Alarms that match the failure
503 Service Unavailable. Page the server-side class. Alarms that match the failure
TLS floor Toward Teams, 1.2 or newer, or the handshake never yields SIP. Alarms that match the failure
SRS down Recording-session alarm. RFC 7866. Retention stays outside the RFC. Alarms that match the failure
Baseline The last good OPTIONS or the current snippet, written down. One change in the window
One variable A single address, codec, or TLS setting. One change in the window
Verify One signal. A 200, or the test call to +15551212. One change in the window
Rollback The previous value of that same variable. One change in the window
Role What the box does on the ladder: UA, proxy, registrar, B2BUA. A glossary tied to a document
B2BUA UAS plus UAC. RFC 7092. Typical SBC. A glossary tied to a document
Document The cite you can open. A logo is not a document. A glossary tied to a document
SDP row RFC 8866 now. RFC 4566 is the one it moved from. A glossary tied to a document
WFM / QM Operations systems. No SIP RFC in this library. Neighbors that are not SIP elements
SIPREC The recording neighbor that is SIP. RFC 7866 and RFC 7865. Neighbors that are not SIP elements
CRM Not a SIP element. No invented vendor features. Neighbors that are not SIP elements
Above the media Analytics and agents. SIP and RTP still carry the session. Neighbors that are not SIP elements
Above RTP Analytics and agents. The packets remain RTP. Workforce tools and agents above the media
Dialog Still Call-ID, CSeq, To tag. RFC 3261. Workforce tools and agents above the media
Disclosed fork SIPREC when the SBC records. RFC 7866. Workforce tools and agents above the media
No invented features Vendor guides or an honest gap. Not a fake RFC. Workforce tools and agents above the media
SSCA SIP School Certified Associate. 20 CEC credits toward BICSI, per the SSCA page. Certifications this edition will name
CLCOR 350-801. Core for CCNP Collaboration and the Collaboration Core specialist. Certifications this edition will name
Concentrations 300-815 CLACC, 300-820 CLHCT, or 300-830 CLCCE. One of them, plus the core. Certifications this edition will name
MS-721 Collaboration Communications Systems Engineer Associate. Certifications change. Use the link. Certifications this edition will name
Banner RFC Editor header beats a blog. When the page and the memory disagree
Cisco page Exam topics change. The Cisco page wins. When the page and the memory disagree
MS-721 link Certifications change. Use the certification page. When the page and the memory disagree
Checked dates February 3, 2026. April 28, 2026. October 29, 2026. Source check 2026-10-14. When the page and the memory disagree
CUCM Registrar and trunk call-control point. CUCM, UCCX, and CVP without a fake script
UCCX Call control through CUCM. Not the carrier trunk. CUCM, UCCX, and CVP without a fake script
CUBE or CUCM Where PSTN SIP lands. CUCM, UCCX, and CVP without a fake script
CVP SIP among CVP, gateways, and CUCM, in that version's guide. CUCM, UCCX, and CVP without a fake script
Local Gateway Premises PSTN pattern for Webex Calling. Webex Calling local gateway, and a separate contact center
Organization PSTN Configured on the Webex Calling PSTN page. Webex Calling local gateway, and a separate contact center
Placeholder host example.com stands in so the lesson does not invent the edge FQDN. Webex Calling local gateway, and a separate contact center
Separate product Webex Contact Center. Check its current design guide. Webex Calling local gateway, and a separate contact center
Call-ID + To tag Dialog match. No To tag on a brand-new INVITE. Signaling drill with the cited answer
Branch Transaction hop. z9hG4bK. Signaling drill with the cited answer
2xx ACK End to end. RFC 6026. Signaling drill with the cited answer
487 INVITE's final response after CANCEL. ACK is hop by hop. Signaling drill with the cited answer
488 This UA. Session description. RFC 3261 and RFC 3264. Media drill with the cited answer
422 RFC 4028. Raise the session interval to Min-SE. Media drill with the cited answer
telephone-event RFC 4733. DTMF in the offer you mean. Media drill with the cited answer
Keying RFC 3711 plus either RFC 4568 or RFC 5764. Record which. Media drill with the cited answer
RFC 7092 B2BUA taxonomy. New Call-ID on the other leg is normal. SBC drill with the cited answer
RFC 5853 SBC deployment requirements. SBC drill with the cited answer
Trunk landing CUBE or CUCM. Not UCCX. SBC drill with the cited answer
Teams pair TLS 1.2+ and an FQDN in Contact. SBC drill with the cited answer
Symptom Silence after connect, Alice at 203.0.113.10. Written first. Symptom, ladder, one packet, one variable
Ladder INVITE, 100, 180, 200, end-to-end ACK. One Call-ID. Symptom, ladder, one packet, one variable
One packet 200 OK SDP. 203.0.113.20, UDP 19642, PCMU. Symptom, ladder, one packet, one variable
One variable Capture that port. Config edits wait, and then they are single. Symptom, ladder, one packet, one variable
Six rows TLS, codecs, numbering, trusted addresses, OPTIONS, SRTP keying. An interconnect checklist, not a config dump
telephone-event On the codec row on purpose. RFC 4733. An interconnect checklist, not a config dump
E.164 Number form. tel URI is RFC 3966. An interconnect checklist, not a config dump
Not a dump Decisions only. No private key, no full dial-peer list. An interconnect checklist, not a config dump
Path CUCM, then a certified SBC, then Teams. No invented SKU. CUCM, a certified SBC, and Teams Direct Routing
CUBE guide Basic SIP configuration for the CUBE case. CUCM, a certified SBC, and Teams Direct Routing
TLS 1.2+ Teams floor. Handshake failures are not SIP codes. CUCM, a certified SBC, and Teams Direct Routing
FQDN Contact sbc.example.com in the lab. An address literal draws 403. CUCM, a certified SBC, and Teams Direct Routing
SDP RFC 4566 moved to RFC 8866. When the RFC number moves
ICE RFC 5245 moved to RFC 8445. Confirm the banner. When the RFC number moves
STUN / TURN RFC 8489 and RFC 8656 in this library. Predecessors 5389 and 5766. When the RFC number moves
Banner A newer obsoleting RFC that this edition does not cite still wins. When the RFC number moves
RFC 8825 Applicability statement for WebRTC. It does not define SIP. Signaling, JSEP, and the media plane
JSEP JavaScript controls signaling. SIP is one allowed choice. Signaling, JSEP, and the media plane
RFC 7118 WebSocket subprotocol sip for SIP messages. Signaling, JSEP, and the media plane
RFC 6455 WebSocket handshake and framing over TCP. Signaling, JSEP, and the media plane
sip WebSocket subprotocol identifier for this transport. RFC 7118: the sip subprotocol
WSS Via transport when the WebSocket used TLS. RFC 7118: the sip subprotocol
transport=ws URI transport parameter added by RFC 7118. RFC 7118: the sip subprotocol
SIP+D2W NAPTR service value for a SIP WebSocket server. RFC 7118: the sip subprotocol
3478 Default STUN port for UDP and TCP. ICE, STUN, and TURN
5349 Default port for STUN over TLS and DTLS. ICE, STUN, and TURN
Host / srflx / relay Local interface, NAT mapping, TURN address. ICE, STUN, and TURN
TURN A controlled relay. One address, multiple peers. ICE, STUN, and TURN
UDP/TLS/RTP/SAVPF Proto token for SAVPF over DTLS with UDP. DTLS-SRTP and the required audio codecs
use_srtp DTLS extension type 14 for SRTP keying. DTLS-SRTP and the required audio codecs
Opus clock a=rtpmap clock 48000 and 2 channels. DTLS-SRTP and the required audio codecs
PCMU and PCMA Required G.711 codecs for WebRTC audio. DTLS-SRTP and the required audio codecs
Step Use the hop buttons. One caption at a time. Browser softphone call flow
101 then INVITE Open the socket before the SIP request. Browser softphone call flow
ACK is not the key SRTP keys come from DTLS, not from ACK. Browser softphone call flow
5737 Example addresses in this ladder are documentation ranges. Browser softphone call flow
Kamailio SIP server. GPLv2+. Registrar, proxy, WebSocket. Four projects, four roles
rtpengine RTP proxy for use with Kamailio. GNU/Linux. Four projects, four roles
Asterisk Open source PBX. PJSIP lives in pjsip.conf. Four projects, four roles
FreeSWITCH SIP profiles, dialplan, media, WebRTC, Event Socket. Four projects, four roles
pjsip.conf Where res_pjsip accounts are written. Asterisk, PJSIP, and the dialplan
extensions.conf The dialplan file named in the overview. Asterisk, PJSIP, and the dialplan
GPL version 2 The license page's stated distribution terms. Asterisk, PJSIP, and the dialplan
ARI Asterisk REST Interface, named next to AMI and AGI. Asterisk, PJSIP, and the dialplan
SIP profile Endpoint configuration the manual teaches. FreeSWITCH profiles, dialplan, and WebRTC
Event Socket Programming interface named in the manual. FreeSWITCH profiles, dialplan, and WebRTC
Two editions Open source, and enterprise from SignalWire. FreeSWITCH profiles, dialplan, and WebRTC
SWML Hosted call-control markup on the SignalWire page. FreeSWITCH profiles, dialplan, and WebRTC
GPLv2+ The license the Kamailio home page states. Kamailio as the SIP server
Registrar A role the page lists, with location service. Kamailio as the SIP server
WebSocket Listed as support for WebRTC. Kamailio as the SIP server
In front of a PBX The page names Asterisk and FreeSWITCH. Kamailio as the SIP server
With Kamailio The README's intended SIP proxy. rtpengine and the integration map
Userspace fallback If the kernel module is unavailable. rtpengine and the integration map
ICE bridge Between ICE-aware and ICE-unaware agents. rtpengine and the integration map
RTP and SRTP The README lists a bridge between them. rtpengine and the integration map
Not legal advice Study notes. The statute and the agency page win. How to read a compliance page
200 OK SIP success. Not a compliance result. How to read a compliance page
Confirm No URL when the link was not opened. How to read a compliance page
Separate planes Voice identity, emergency location, and SMS campaigns are different documents. How to read a compliance page
Identity SIP header for the STIR signature. RFC 8224. STIR and SHAKEN
PASSporT Signed token. RFC 8225. STIR and SHAKEN
SHAKEN Industry mechanism described on the STI-GA page. STIR and SHAKEN
toKENs The STI-GA page writes SHAKEN out with that word. STIR and SHAKEN
PL 115-127 Kari's Law. Direct 911 dialing from MLTS. Kari's Law, RAY BAUM'S Act, and SIP location
Section 506 Dispatchable location proceeding, including MLTS. Kari's Law, RAY BAUM'S Act, and SIP location
RFC 6442 How SIP conveys location. Kari's Law, RAY BAUM'S Act, and SIP location
RFC 6881 Emergency calling BCP. Not a statute. Kari's Law, RAY BAUM'S Act, and SIP location
10DLC A2P messaging channel on the registry page. 10DLC is a messaging channel
CSP Campaign service provider. The page says to be one before registering. 10DLC is a messaging channel
Campaign ID Given to the connectivity partner, on that page. 10DLC is a messaging channel
Not SIP Do not add it to a voice INVITE and call it done. 10DLC is a messaging channel
Sort first SIP, media, identity, location, or messaging. Troubleshooting without a legal conclusion
No live 911 Use the provider test. This lab is not that test. Troubleshooting without a legal conclusion
Identity header RFC 8224. Presence is not a legal conclusion. Troubleshooting without a legal conclusion
Not advice Counsel and the current rule outrank the quiz. Troubleshooting without a legal conclusion
Citation The glossary row's authority. Glossary and the mind map
Mind map Topics. Not a network. Glossary and the mind map
Confirm No link when the URL was not opened. Glossary and the mind map
Next track Use the map to pick WebRTC, the open core, or compliance. Glossary and the mind map
SIPREC Fork a session to a recorder. RFC 7866. Sister technologies around the session
WFM Schedules. SIP only shows the symptom, such as 480. Sister technologies around the session
CRM Events and numbers. Not a proxy. Sister technologies around the session
Twilio docs API reference and tutorials. Confirm product names there. Sister technologies around the session
pjsip.conf Asterisk SIP resources. Configuration and troubleshooting seams
extensions.conf Asterisk dialplan. Configuration and troubleshooting seams
101 first No WebSocket, no SIP. Configuration and troubleshooting seams
Plane Name signaling, media, identity, location, or messaging. Configuration and troubleshooting seams
CUCM trunk SIP toward the SBC. Profile is CUCM's object. Bridges from enterprise UC to open platforms
Local Gateway Webex Calling premises PSTN role. Bridges from enterprise UC to open platforms
Open side Kamailio, Asterisk, or FreeSWITCH, per their pages. Bridges from enterprise UC to open platforms
Not inside CUCM The open dialplan stays on the open server. Bridges from enterprise UC to open platforms
ARI Asterisk REST Interface, named with AMI and AGI. Integration hooks that are already named
Event Socket FreeSWITCH programming path in the manual. Integration hooks that are already named
SIPREC Recording hook. RFC 7866. Integration hooks that are already named
No live connector This repository does not run the PBX or the registry. Integration hooks that are already named

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RFC 3261 — last checked 2026-10

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