Post-dial delay and slow call setup

See also

Sources

1. 1
   Cisco's tech note explains that when a dialed DN also partially matches a longer route pattern such as an international pattern, Unified CM waits for the interdigit timer to expire before routing to the DN.

   Urgent Priority Configuration on Directory Number · Problem description / example section · Checked 2026-10-14
2. 2
   Cisco documents the Unified CM T302 interdigit timer default as 15 seconds.

   Urgent Priority Configuration on Directory Number · Background / problem description section · Checked 2026-10-14
3. 3
   In Cisco Unified CM the interdigit timer is the T302 service parameter under Clusterwide Parameters (Device - General).

   Urgent Priority Configuration on Directory Number · Background / problem description section · Checked 2026-10-14
4. 4
   With Urgent Priority enabled on the Directory Number Configuration page, Unified CM routes the call at once to the fully qualified DN instead of waiting on T302; the note says this DN option arrived with CUCM 10.0.

   Urgent Priority Configuration on Directory Number · Configuration section, Urgent Priority on Directory Number · Checked 2026-10-14
5. 5
   When the first SRV target is silent rather than refusing, a client following RFC 3263 only moves to the next target after a transaction timeout, so each dead target can add up to one Timer B interval of post-dial delay unless the implementation shortens it.inferred

   RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Section 4.3 Details of RFC 2782 Process · Checked 2026-10-14
6. 6
   RFC 3263 says the identity of an available server would ideally be cached for some time to reduce call setup delay on subsequent calls.

   RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Section 2 Problems DNS is Needed to Solve · Checked 2026-10-14
7. 7
   RFC 3263 treats a 503, a transport failure, or a Timer B or F timeout as failure and has the client send a new request with a new Via branch to the next element in the SRV-ordered list.

   RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Section 4.3 Details of RFC 2782 Process · Checked 2026-10-14
8. 8
   RFC 3263 has a SIP client resolve a domain with NAPTR first, then SRV for the chosen transport, and falls back to A or AAAA lookups when no SRV records are found.

   RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Sections 4.1 Selecting a Transport Protocol and 4.2 Determining Port and IP Address · Checked 2026-10-14
9. 9
   Under RFC 3263 a URI with an explicit transport goes straight to an SRV query, and a numeric IP target is used directly with no NAPTR or SRV lookup.

   RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Sections 4.1 and 4.2 · Checked 2026-10-14
10. 10
    RFC 3263 notes that because several hops may each perform DNS lookups, the time available for DNS lookups at each hop is limited.

    RFC 3263: Session Initiation Protocol (SIP): Locating SIP Servers · Section 2 Problems DNS is Needed to Solve · Checked 2026-10-14
11. 11
    ITU-T E.721 (05/99), Network grade of service parameters and target values for circuit-switched services in the evolving ISDN, is the edition in force, superseding the 11/88 and 08/91 editions.

    E.721: Network grade of service parameters and target values for circuit-switched services in the evolving ISDN · Recommendation page edition list · Checked 2026-10-14
12. 12
    On Cisco IOS 12.3T voice ports the timeouts interdigit command defaults to 10 seconds with a range of 0 to 120 seconds, and 0 disables the timer.

    Cisco IOS Voice Command Reference - Commands: timeout (auto-config application) through timing clear-wait [Cisco IOS Software Releases 12.3 T] · Command entry timeouts interdigit (voice port) · Checked 2026-10-14
13. 13
    On Cisco IOS voice ports the interdigit timer restarts with each digit until the destination address is identified, and if it expires first the caller hears a tone and the call is terminated.

    Cisco IOS Voice Command Reference - Commands: timeout (auto-config application) through timing clear-wait [Cisco IOS Software Releases 12.3 T] · Command entry timeouts interdigit (voice port), Usage Guidelines · Checked 2026-10-14
14. 14
    On Cisco IOS 12.3T the sip-ua retry invite command defaults to 6 retries with a range of 1 to 10.

    Cisco IOS Voice Command Reference - Commands: R [Cisco IOS Software Releases 12.3 T] · Command entry retry invite · Checked 2026-10-14
15. 15
    Lowering sip-ua retry invite and timers trying on a Cisco gateway shortens how long an INVITE to a silent next hop waits before the gateway hunts to the next dial peer, which shortens post-dial delay during trunk failure.inferred

    Cisco IOS Voice Command Reference - Commands: R [Cisco IOS Software Releases 12.3 T] · Command entry retry invite (with timers trying entry in cisco-ios-vcr-123t-timeout-to-timing) · Checked 2026-10-14
16. 16
    On Cisco IOS 12.3T the sip-ua timers trying setting, the wait for a 100 response to an INVITE, defaults to 500 ms with a range of 100 to 1000 ms.

    Cisco IOS Voice Command Reference - Commands: timeout (auto-config application) through timing clear-wait [Cisco IOS Software Releases 12.3 T] · Command entry timers (sip-ua) · Checked 2026-10-14
17. 17
    RFC 3578 names converting PSTN overlap signalling into SIP en-bloc signalling at the gateway as the preferred solution.

    RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP) · Section 1 Introduction; Section 2 · Checked 2026-10-14
18. 18
    RFC 3578 states that converting overlap signalling to en-bloc sometimes results in unacceptable call setup delays of multiple seconds for human users.

    RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP) · Section 1 Introduction · Checked 2026-10-14
19. 19
    When overlap signalling is carried into SIP under RFC 3578, each new set of digits produces a new INVITE with the same Call-ID and an updated Request-URI, and earlier INVITEs are typically ended by 484 responses.

    RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP) · Sections 3 and 3.2 · Checked 2026-10-14
20. 20
    RFC 3578 says overlap signalling in SIP is unsuitable where one valid number is a prefix of another valid number.

    RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP) · Section 3.1 · Checked 2026-10-14
21. 21
    Under RFC 3578 en-bloc conversion, once the minimum digits are present the gateway collects digits until timer T10 expires or a stop digit arrives, then sends one INVITE with the digits collected so far.

    RFC 3578: Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP) · Section 2.2 Receiving the minimum number of digits · Checked 2026-10-14
22. 22
    RFC 6076 does not appear to use the term post-dial delay; practitioners mapping PDD to an IETF metric are mapping it to SRD by analogy.inferred

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Whole document; Section 4.3 is the closest related text · Checked 2026-10-14
23. 23
    In SIP, 100 Trying indicates the request was received by the next-hop server, and a stateful proxy does not forward 100 Trying upstream.

    RFC 3261 — SIP: Session Initiation Protocol · Section 21.1.1 100 Trying; Section 16.7 step 5 · Checked 2026-10-14
24. 24
    An INVITE server transaction must generate a 100 Trying unless it knows the TU will generate a provisional or final response within 200 ms.

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.2.1 INVITE Server Transaction · Checked 2026-10-14
25. 25
    In SIP, 180 Ringing means the UA receiving the INVITE is trying to alert the user, and the response may be used to initiate local ringback.

    RFC 3261 — SIP: Session Initiation Protocol · Section 21.1.2 180 Ringing · Checked 2026-10-14
26. 26
    In SIP, 183 Session Progress conveys call progress information that is not otherwise classified.

    RFC 3261 — SIP: Session Initiation Protocol · Section 21.1.5 183 Session Progress · Checked 2026-10-14
27. 27
    After a SIP INVITE client transaction receives a 1xx response, INVITE retransmissions cease and the client waits for further responses.

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.1.1.1 Overview of INVITE Transaction · Checked 2026-10-14
28. 28
    SIP 484 Address Incomplete means the server received a request whose Request-URI was incomplete.

    RFC 3261 — SIP: Session Initiation Protocol · Section 21.4.22 484 Address Incomplete · Checked 2026-10-14
29. 29
    A SIP proxy or UAC receiving 503 Service Unavailable should attempt to forward the request to an alternate server, and a 503 without Retry-After must be treated as a 500.

    RFC 3261 — SIP: Session Initiation Protocol · Section 21.5.4 503 Service Unavailable · Checked 2026-10-14
30. 30
    In SIP, T1 is an estimate of the round-trip time and defaults to 500 ms.

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.1.1.1 Overview of INVITE Transaction · Checked 2026-10-14
31. 31
    Over an unreliable transport the SIP INVITE client transaction starts Timer A at T1 and retransmits the INVITE at intervals that double after each transmission.

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.1.1.2 Formal Description · Checked 2026-10-14
32. 32
    For any transport the SIP INVITE client transaction starts Timer B at 64*T1, and if Timer B fires while still in the Calling state the transaction should inform the TU that a timeout occurred.

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.1.1.2 Formal Description · Checked 2026-10-14
33. 33
    RFC 6076 says that when no provisional response is received and a 200 is the first status message, the 200 ends the successful SRD interval.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3.1 Successful Session Setup SRD · Checked 2026-10-14
34. 34
    For a failed session request, RFC 6076 SRD ends at the first provisional or failure response, where failure means 4xx other than 401 402 and 407, 5xx, or possibly 6xx.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3.2 Failed Session Setup SRD · Checked 2026-10-14
35. 35
    RFC 6076 says that when a forked INVITE creates multiple dialogs an SRD measurement should be taken for each early or confirmed dialog created.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 5.4 Forking · Checked 2026-10-14
36. 36
    RFC 6076 defines Session Request Delay (SRD) as the time of the status-indicative response minus the time of the INVITE, expressed in seconds.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3 Session Request Delay · Checked 2026-10-14
37. 37
    RFC 6076 states that its metrics are inclusive of the time spent on authentication exchanges, so digest challenges add to measured SRD.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 5.3 Authentication · Checked 2026-10-14
38. 38
    RFC 6076 requires that SRD results for failed session requests are not combined in the same result with successful requests.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3 Session Request Delay · Checked 2026-10-14
39. 39
    RFC 6076 states that SRD is measured at the originating user agent only.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3 Session Request Delay · Checked 2026-10-14
40. 40
    RFC 6076 notes that using PRACK (RFC 3262) may improve the quality and consistency of SRD results.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3.1 Successful Session Setup SRD · Checked 2026-10-14
41. 41
    RFC 6076 states that the start time t1 is not reset when the user agent retransmits the same message within the same transaction, so retransmission waits count toward SRD.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4 Metrics (general measurement rules) · Checked 2026-10-14
42. 42
    RFC 6076 describes SRD as similar to the Post-Selection Delay defined in ITU-T E.721.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3 Session Request Delay · Checked 2026-10-14
43. 43
    RFC 6076 SRD examples with a 302 redirect measure from the original INVITE to the 180 or failure response from the redirected-to user agent, spanning the redirection.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Sections 4.3.1 and 4.3.2 redirect example figures · Checked 2026-10-14
44. 44
    For a successful session request, RFC 6076 SRD runs from the first bit of the initial INVITE to the last bit of the first non-100 provisional response.

    RFC 6076: Basic Telephony SIP End-to-End Performance Metrics · Section 4.3.1 Successful Session Setup SRD · Checked 2026-10-14
45. 45
    An SBC ACL that admits only some Microsoft Direct Routing datacenter addresses is a plausible cause of intermittent slow outbound call setup, because refused connections trigger retries from other datacenters before the call proceeds.inferred

    Trunk failover on outbound calls - Microsoft Teams · Section Failover on network errors · Checked 2026-10-14
46. 46
    Teams Direct Routing FailoverResponseCodes defaults to 408, 503 and 504, and a matching final response from the SBC to an initial INVITE makes Direct Routing try another SBC in the voice routing policy.

    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, row Failover response codes / FailoverResponseCodes · Checked 2026-10-14
47. 47
    Microsoft warns that in organizations with slow networks and gateway responses the FailoverTimeSeconds setting could result in calls being dropped unnecessarily.

    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, row Failover times (seconds) / FailoverTimeSeconds · Checked 2026-10-14
48. 48
    With FailoverTimeSeconds set, Teams Direct Routing routes outbound calls not answered by the gateway within that time to the next available trunk, and drops the call if no further trunk exists.

    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, row Failover times (seconds) / FailoverTimeSeconds · Checked 2026-10-14
49. 49
    In Teams Direct Routing the FailoverTimeSeconds SBC setting defaults to 10 seconds.

    Connect your Session Border Controller (SBC) to Direct Routing · SBC settings table, row Failover times (seconds) / FailoverTimeSeconds · Checked 2026-10-14
50. 50
    If a Teams Direct Routing trunk cannot be connected because of a refused connection, TLS timeout or other network-level issue, the same trunk is retried from a different Microsoft datacenter, possibly in another region.

    Trunk failover on outbound calls - Microsoft Teams · Section Failover on network errors · Checked 2026-10-14
51. 51
    Teams Direct Routing fails over on configured SIP codes only when the SBC has sent no non-100 provisional response, to prevent double ringing in the telephony network.

    Trunk failover on outbound calls - Microsoft Teams · Section Failover of specific SIP codes received from the Session Border Controller (SBC) · Checked 2026-10-14
52. 52
    With the default T1 of 500 ms, an INVITE that draws no response at all is abandoned only after about 32 seconds, which is far longer than callers tolerate before hanging up.inferred

    RFC 3261 — SIP: Session Initiation Protocol · Section 17.1.1.2 Formal Description (Timer B = 64*T1) combined with Section 17.1.1.1 (T1 = 500 ms) · Checked 2026-10-14
53. 53
    Twilio Voice Insights reports High PDD as a percentage of calls, compared against the 99th percentile of all Twilio calls to the destination country.

    Voice Insights Call Insights Dashboard · Connection Rate section, metrics table, High PDD row · Checked 2026-10-14
54. 54
    Twilio Voice Insights defines PDD as the number of seconds elapsed between dialing the last digit of the phone number and the start of ringing.

    Voice Insights Call Insights Dashboard · Connection Rate section, metrics table, Post-dial-delay row · Checked 2026-10-14
55. 55
    Twilio's glossary attributes most post-dial delay to improper SIP signalling responses from destination networks.

    What is Post-Dial Delay? · Glossary entry What is Post-Dial Delay?, causes paragraph · Checked 2026-10-14
56. 56
    Twilio's glossary defines post-dial delay technically as the delta between the SIP INVITE and a 180 Ringing or 183 Session Progress response.

    What is Post-Dial Delay? · Glossary entry What is Post-Dial Delay?, definition paragraph · Checked 2026-10-14
57. 57
    Twilio's glossary says most North American users notice post-dial delay of about 5 seconds, while in some locations such as South Africa 10 seconds or more is common.

    What is Post-Dial Delay? · Glossary entry What is Post-Dial Delay?, tolerance paragraph · Checked 2026-10-14
58. 58
    Twilio's glossary says geographically remote destinations and non-geographic destinations such as satellite phones are prone to greater post-dial delay.

    What is Post-Dial Delay? · Glossary entry What is Post-Dial Delay?, causes paragraph · Checked 2026-10-14
59. 59
    Twilio suggests 4 seconds as a starting timeout interval for a PBX or SBC to stop waiting on one resolved Elastic SIP Trunking termination IP and try the next.

    Programmable Voice Failover Best Practices · Elastic SIP Trunking Termination, subsection Utilize all IP addresses for a termination SIP URI · Checked 2026-10-14
60. 60
    Twilio says resolving a termination SIP URI FQDN returns 3 to 4 IP addresses across availability zones and advises using all of them rather than pinning one.

    Programmable Voice Failover Best Practices · Elastic SIP Trunking Termination, subsection Utilize all IP addresses for a termination SIP URI · Checked 2026-10-14

Documents

Cite this page

WarmTransfer. (2026, October 14). Post-dial delay and slow call setup. WarmTransfer. https://warmtransfer.net/knowledge/post-dial-delay

@misc{warmtransfer-post-dial-delay,
  title  = {Post-dial delay and slow call setup},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/post-dial-delay},
  note   = {Verified 2026-10-14}
}