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Overview

The

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relays the INVITE requests it receives. Egressing INVITE may/may not have SDP depending on presence of SDP in the received INVITE and late offer interworking configuration.
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inserts Allow header containing the following methods: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, MESSAGE, PUBLISH and OPTIONS. There are configurable options for the support of REFER, SUBSCRIBE, NOTIFY, OPTIONS, MESSAGE, PUBLISH and INFO, and when these methods are configured as not supported, these methods are not added in the Allow header.

The

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adds a Supported header containing the following extensions: "100rel", "timer" and "path" extensions. Addition of “100rel”, “timer” and “path” extensions is configurable. As a part of RCS support,
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relays “Require” header with “recipient-list-invite” option-tag.

The

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can be configured to transparently pass the “From’ and “To” headers from ingress to egress call-leg. Additionally, for access deployments, the 
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can be configured to use Username in the To header instead of the Request-URI, while terminating calls to IP-PBX.

The

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supports processing or relaying the ‘replaces header’ received in the incoming SIP request. This done based on the ‘Relay Replaces Header flag’ in SIP service on the
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. If the 
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is configured to process INVITE with Replaces locally and if the dialog doesn’t exist, the INVITE is rejected with 481 response. the
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 processes INVITE with Replaces locally only on confirmed dialogs.

The

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supports reception of INVITE with and without SDP Offer.

  • If the INVITE is received with SDP offer, the 
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    sends the SDP Answer in the first reliable provisional response or in success final response.
  • If the initial INVITE does not contain an SDP offer, the
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    sends the SDP offer in the first reliable 18x or 200 OK response and the answer is expected in PRACK or ACK method (asappropriate).
Even though the
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sends Allow and Supported headers by default in the outgoing INVITE, it does not discard an incoming INVITE if these headers are NOT present.

INVITE Support

The

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supports sending SIP INVITE requests to establish VoIP calls. When a User Agent Client (UAC) desires to initiate a session (for example, audio, video, or a game), it formulates an INVITE request. The INVITE request asks a server to establish a session. This request may be forwarded by proxies, eventually arriving at one or more User Agent Servers (UASs) that can potentially accept the invitation. These UASs will frequently need to query the user about whether to accept the invitation. After some time, those UASs can accept the invitation (meaning the session is to be established) by sending a 2xx response.

If the invitation is not accepted, a 3xx, 4xx, 5xx or 6xx response is sent, depending on the reason for the rejection. Before sending a final response, the UAS can also send provisional responses (1xx) to advise the UAC of progress in contacting the called user.

Sending an INVITE request within an existing dialog to change addresses or ports, add or delete a media stream, and so forth is known as a re-INVITE.

Do not use deriveFromOtherLeg when configuring H.323 or SIP trunk groups to use INVITEs with no SDPs.

reINVITE Handling

Reference: RFC 6141

The

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does not provide any interworking between an error response and 200 (OK) for a reINVITE.

reINVITE Handling  by UAS

No state changes are performed in the 

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if a reINVITE is rejected by the downstream entity.

The

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supports the following UAS behavior for a call which is set up with only an audio stream:

  • When a reINVITE is received adding a video stream (SDP has both audio and video components), and if the reINVITE is rejected by the downstream entity, the audio stream remains intact and is still active.

    Call Flow Example

  • When a reINVITE is received adding a video stream and also updating the IP address for the audio stream (SDP has both audio and video components), and if the corresponding reINVITE on the other leg is rejected by the downstream entity, the 

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    continues to use the old IP address for the audio stream.

    Call Flow Example

  • When a reINVITE is received adding a video stream and also updating the IP address for the audio component (SDP has both audio and video components), and if the corresponding reINVITE on the other leg is replied with 200 (OK), but video component has port=0. The 

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    replies with 200 (OK), video stream is not created and the new IP address is used for audio.

    Call Flow Example

  • When the
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    receives a reINVITE to add a video stream but does not have enough bandwidth to do so, the 
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    generates a 200 (OK) response with port=0 for video stream.

  • The
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    does not propagate a reINVITE/UPDATE to the other leg if it does not change any characteristics (e.g. codec, digit transfer mode) in the existing streams on the other leg.

  • When an egress server changes the SDP from 18x message to 200 OK message (which is a violation of RFC 3261), the 
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    interworks by sending the last sent SDP in 200 OK and then sends a RE-INVITE to correct the SDP. This Re-INVITE can cause problems if the INVITE/Re-INVITE is challenged for authentication or because the endpoints change Secure Real-Time Transport Protocol (SRTP) keys for every new SDP offer. In these scenarios, the 
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     updates the SDP in 200 OK itself.

UAC Behavior

The

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does not generate a reINVITE/UPDATE upon receipt of an error response for a reINVITE. It is assumed that downstream entity is RFC6131 compliant and keeps session characteristics the same as before the rejected reINVITE transaction.

Glare Situations

The

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replies with 491 (Request Pending) if a reINVITE is received while a previous one is still pending.

 

Clarifications on Canceling re-INVITEs

When canceling reINVITES, the

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always replies with 487 (Request Terminated).

Refreshing a Dialog’s Target and its Subsections

The

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 supports refreshing a dialog's target and its subsections as well as updating remote target based on an unreliable response.

 

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