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| Transparency_Profile_Note |
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| Transparency_Profile_Note |
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SDP Transparency
Session Description Protocol (SDP) is a set of rules defining how multimedia sessions are set up to allow all end points to effectively participate in the session.
Interactive Connectivity Establishment (ICE) is a protocol used by systems that cannot determine their own transport address as seen from the remote end, but that can provide several possible alternatives (see ICE-Lite Support below). The SDP transparency feature allows SDP and ICE media to pass through the
as-is to the far end.
SDP Transparency functionality includes:
- All unknown SDP attributes are transparently passed.
- All unknown components of known SDP attributes are dropped.
- Any unknown audio codecs are dropped.
- All known and unknown video codecs are transparently passed.
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When SDP Transparency is enabled, all IP Signaling Profile SDP-related flags are overridden. |
For additional audio/video support topics, refer to:
Suppressing 183 Response Without SDP
The
supports suppressing the 183 response without SDP upon receipt of 3xx Redirect response for reasons such as:
An endpoint perceives a 183 without SDP response as being an extra message resulting in unexpected behavior.
Endpoints that behave irregularly upon receipt of a 183 without SDP response prior to cut-through.
Two IP Signaling Profile ingress flags are available to configure the
to suppress 183 response without SDP:
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language | none |
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title | CLI Syntax |
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% set profiles signaling ipSignalingProfile <SIP profile name> ingressIpAttributes flags suppress183For3xxRedirectResponsek <disable | enable>
% set profiles signaling ipSignalingProfile <SIP profile name> ingressIpAttributes flags suppress183WithoutSdp <disable | enable>
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The following table describes the effects of enabling/disabling these flags.
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0 | Table |
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1 | Suppress 183 Response Without SDP Flags |
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Suppress 183 for 3xx Redirect Response | Suppress 183 without SDP | Action |
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Enabled | Enabled | If 183 without SDP is due to 3xx redirect response, it is suppressed due to “Suppress 183 for 3xx Redirect Response” flag. If 183 without SDP is triggered for reason other than 3xx redirect response, it is suppressed due to “Suppress 183 without SDP ” flag.
| Enabled | Disabled | 3xx redirect response triggers 183 without SDP, and hence is suppressed | Disabled | Enabled | A '183 without SDP triggered due to 3xx redirect response' (or for any other reason) is suppressed. | Disabled | Disabled | Existing behavior persists. |
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For configuration details, refer to Ingress IP Attributes - SIP - CLI or Ip Signaling Profile - Ingress Ip Attributes (EMA).
Passing Audio Codecs in SDP Offer
The
supports passing the list of received audio codecs in the SDP offer to PSX in the policy request. The
also passes the received audio codec information list as received in the ingress SDP to PSX in the policy request.
The
uses the modified calling number returned by the PSX in the policy response in formatting the egress call leg request. The P-CDR information is written onto the
accounting records,
populating the “Egress External Accounting Data” field for STOP and ATTEMPT CDR records. Note |
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This feature is not applicable when the is configured for ERE mode. |
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| DeriveFromOtherLeg limitation |
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| DeriveFromOtherLeg limitation |
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Excerpt |
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The sendAllAllowedCodecsForLateMediaInviteOrReInvite flag controls the handling of audio codecs that the offers in response to a late media INVITE or Re-INVITE without SDP for transcoded calls.- When this flag is disabled for transcoded calls (default behavior), the offers the codec used for transcoding on the leg.
- When the flag is enabled for transcoded calls, the offers multiple codecs which include:
- The subset of the codecs that the associated peer supports.
- The transcoded codecs that the associated DSP channel supports. This includes the codec currently being used for transcoding.
For pass-through calls, the always offers a subset of the codecs advertised by the associated peer. The Offer codec list is classified as a subset because the list can be subjected to codec policy filters. |
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Example 1: G711 pass-through call
- Ingress Offer: G711, G729
- Egress Offer: G711, G729, AMR, G726
- Ingress Answer: G711
- Egress Answer: G711
If a re-INVITE with no SDP is received on egress, the
generates an offer for G711, G729. The G726 and AMR are not offered because the call is marked as a relay call with DSP removed, and hence the transcoded codecs.
Example 2: G711 to AMR transcoded call
- Ingress Offer: G711, G729
- Egress Offer: G711,G729, AMR, G726
- Egress Answer: AMR
- Ingress Answer: G711
If a re-INVITE with no SDP is received on egress, by default, the
generates an offer for AMR. If the “Send All Allowed Codecs For Late Media Invite Or Re-Invite” flag is enabled then the
generates a codec list of G711,G729, AMR, and G726 in the Offer.
Refer to Common IP Attributes - SIP - CLI or Common Ip Attributes - Flags (EMA) for configuration details.
ANAT Support
Alternative Network Address Types (ANAT) semantics for the SDP grouping framework allows the expression of alternative network addresses (e.g. different IP versions) for a particular media stream. This ability is useful in environments with both IPv4-only hosts and IPv6-only hosts.
The
supports ANAT formatting within the SDP offer to facilitate both an IPv4 and IPv6 address types. In addition, the
allows a configured address type preference whereby the user can configure which IP version takes precedence when multiple IP version types are supported. If the
receives an ANAT offer and only a single IP version is supported on the received interface, that IP version is used regardless of the configured IP version type preference. Refer to
SIP Trunk Group - Media - CLI or
SIP Trunk Group - Media for configuration details.
ICE-Lite Support
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| ICE-Lite_Support |
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| ICE-Lite_Support |
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Bundling of Streams
supports bundling of various streams (audio/video) over the same IP/port pair using a SDP grouping framework extension named
BUNDLE. The
BUNDLE is used with the SDP OFFER/ANSWER mechanism to negotiate the usage of a single
address:port combination (BUNDLE address) for receiving media (bundled media) associated with multiple SDP media streams (
"m=" lines ). The
address:port combination used for sending bundled media may be the same as the BUNDLE address, used to receive bundled media, depending on whether symmetric RTP is used. The
sdpAttributesSelectiveRelay
flag must be enabled for bundling.
The SDP media-level attribute, "bundle-only", is parsed and used to identify that specific media is only used if bundled, and resides within a BUNDLE group. The "bundle-only" attribute is a property attribute with no value.
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The can accept a mixture of bundled and unbundled streams, but currently only single bundle is supported. |
The use of a BUNDLE group and a BUNDLE address also allows the usage of a single set of Interactive Connectivity Establishment (ICE) candidates for multiple "m=" lines.
If RTP/RTCP multiplexing is used, the same address:port combination is used for all RTP and RTCP packets.
Single-bundle
The
accepts offers that express bundling with a single bundle. A bundle may have one or more streams.
Stream
- A stream can be tagged as bundle-only and have a port set to 0.
- The policers for the stream which represents the bundle reflect the total bandwidth for all streams within the bundle.
- Only relay scenarios are supported, any inter-working causes the streams to unbundle.
Bundle Only
The
accepts offers that include bundled streams with the bundle-only attribute set.
- Offers with "m=" lines tagged as bundle-only with the port set to 0.
- Non-compliant offers which associate bundle-only with a non-zero port, the "m=" line is also accepted; the stream is treated as a bundle-only stream.
- Media attributes received with a bundle-only stream offer are handled considering that the stream is present and output in the outbound offer. (media codecs, fmtp options)
Bundle Group
The bundling of streams is negotiated using the "a=group:BUNDLE" attribute at the session level:
- The OFFER contains the BUNDLE attribute followed by a list of streams which are part of the BUNDLE.
- The OFFER contains unique address:port combinations for each stream as the peer may not support bundling the streams.
If the peer is able to support the bundle:
- The ANSWER contains the BUNDLE attribute.
- The ANSWER contains the same address:port combination for all streams within the bundle.
OFFER and ANSWER
The following is an example for OFFER and ANSWER mechanism.
OFFER
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v=0
o=alice 2890844526 2890844526 IN IP4 10.32.241.3
s=
c=IN IP4 10.32.241.3
t=0 0
a=group:BUNDLE audio video
m=audio 10000 RTP/AVP 0
a=mid:audio
a=rtpmap:0 PCMU/8000
a=extmap 1 urn:ietf:params:rtp-hdrext:sdes:mid
m=video 10002 RTP/AVP 31 32
a=mid:video
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
a=extmap 1 urn:ietf:params:rtp-hdrext:sdes:mid
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ANSWER
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v=0
o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
a=group:BUNDLE audio video
m=audio 20000 RTP/AVP 0
a=mid:audio
a=rtpmap:0 PCMU/8000
a=extmap 1 urn:ietf:params:rtp-hdrext:sdes:mid
m=video 20000 RTP/AVP 32
a=mid:video
a=rtpmap:32 MPV/90000
a=extmap 1 urn:ietf:params:rtp-hdrext:sdes:mid
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Interactions with Security
Bundling is only allowed when security is configured to be relayed, and:
- DTLS-SRTP (and DTLS-SCTP for data) is relayed.
- SDES SRTP is relayed
Limitations
- Bundling can only be relayed and not inter-worked.
- Configuration for LI or a call requiring transcoding, DTMF or security termination is unbundled.
- Bundling is not supported over the GW protocol.
RTP and RTCP Multiplexing
The
allows relaying multiplexed RTP and RTCP traffic.
Within a BUNDLE group, the offerer and answerer must enable RTP/RTCP multiplexing for the RTP-based media associated with the BUNDLE group.
When RTP/RTCP multiplexing is enabled, the same address:port combination is used for sending all RTP and RTCP packets associated with the BUNDLE group. Each endpoint sends the packets towards the BUNDLE address of the other endpoint. The same address:port combination may be used for receiving RTP and RTCP packets.
- The allows relay of multiplexed RTP and RTCP traffic only when the rtcp-mux is supported on both sides of the call.
- The "a=rtcp-mux" attribute negotiates multiplexing when RTCP is enabled on both sides of the call.
- The rtcp-mux attribute takes precedence over the "a=rtcp:" attribute when it has to accept SDPs from the peer.
- When rtcp-mux is used within a bundle it must be enabled for all streams in a bundled or disabled for all.
Interactions with ICE
Using bundling of media streams and the support of RTCP multiplexing ensures that only a single ICE transaction on one port is required. This limits the occurrence of NAT traversal issues and reduces the time needed to establish a call.
For the OFFER:
- The ICE data is unique for each stream.
- A candidate is included for RTCP in the initial OFFER and for RTP.
- A bundle-only "m=" line does not include ICE candidate or ufrag/pwd.
For the ANSWER:
- The ICE data matches the chosen master stream.
- A single ICE candidate line for the RTP component is only included.
RTCP Multiplexing to Non-RTCP Multiplexing Inter-working
The RTP contains two components:
- a data transfer protocol and
- an associated Real-time Control Protocol (RTCP).
With the increased use of Network Address Port Translation (NAPT), using separate UDP ports to transport RTP and RTCP has become problematic as maintaining multiple NAT bindings can be costly. To overcome these scenarios, the
supports RTP and RTCP flows on a single port to ease NAT traversal. This functionality is known as RTP and RTCP Multiplexing or RTCP-Mux.
Prior to 6.1.0 release, the
supported RTCP-Mux in pass-through scenarios where the
was relaying media end to end. This support was sufficient for some limited use cases, such as WRTC-to-WRTC call scenarios where the endpoints terminated media encryption with no need for the
to inter-work with any media attributes.
The
is enhanced to support inter-working between RTCP-Mux to non-RTCP-Mux as well as in the reverse direction. In this scenario, both egress and ingress packet Service Profiles (PSPs) either support RTCP-Mux or fallback to transport RTP and RTCP streams using separate ports. The
rtcpMux
flag is added to the PSP to configure RTCP-Mux.
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When RTP interception is triggered on an RTCP-Mux call (for example, lawful intercept), the forks the RTP and RTCP streams to a single port towards the mediation server. If the intercept is performed in an RTCP-Mux session, the stream sent towards the mediation server is also multiplexed with RTP and RTCP sent over the same port. |
The
supports RTCP-Mux for the following features/functionaltiy:
- Multimedia calls
- Monitoring calls
- Both direct media and OMR call scenarios
- DTLS and SRTP scenarios (irrespective of whether the encryption protocol terminates at the or is relayed transparently end-to-end)
- Multiplexing RTCP and inter-working with non-RTCP Mux for both pass-through (with or without "RTCP termination" enabled) and transcoded calls
- Bundled streams for WRTC calls
- Tones and Announcements
- Using SRTP and SRTCP (This is supported in both the cases of SRTP pass-through and SRTP to RTP inter-working.)
RTCP bandwidth requirements do not change with the use of RTCP-Mux. The
transmits bandwidth attributes (RR/RS) as per the existing configuration.
Info |
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The rtcpMux flag is visible only when the rtcp control is enabled on the PSP. |
The following table describes the RTCP-Mux to non-RTCP-Mux inter-working scenarios for offer/answer:
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1 | RTCP-Mux to non-RTCP-Mux Interworking |
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Inter-working Scenarios | Offer from Ingress | Offer to Egress | Answer from Egress | Answer to Ingress | RTCP Flow Ingress Port(s) → Egress Port(s) |
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RTCP-Mux enabled on both PSPs | “a=rtcp-mux" | “a=rtcp-mux” | “a=rtcp-mux” | “a=rtcp-mux” | (P1)→(P1) | “a=rtcp-mux” | “a=rtcp-mux" | No Mux | “a=rtcp-mux” | (P1)→(P1,P2) | No Mux | “a=rtcp-mux” | “a=rtcp-mux” | No Mux | (P1,P2)→(P1) | No Mux | “a=rtcp-mux” | No Mux | No Mux | (P1,P2)→(P1,P2) | RTCP-Mux enabled on ingress only | “a=rtcp-mux” | No Mux | No Mux | “a=rtcp-mux” | (P1)→(P1,P2) | No Mux | No Mux | No Mux | No Mux | (P1,P2)→(P1,P2) | RTCP-Mux enabled on egress only | “a=rtcp-mux” | “a=rtcp-mux” | “a=rtcp-mux” | No Mux | (P1,P2)→(P1) | “a=rtcp-mux” | “a=rtcp-mux” | No Mux | No Mux | (P1,P2)→(P1,P2) | No Mux | “a=rtcp-mux” | “a=rtcp-mux” | No Mux | (P1,P2)→(P1) | No Mux | “a=rtcp-mux” | No Mux | No Mux | (P1,P2)→(P1,P2) | RTCP-Mux disabled on both PSPs | “a=rtcp-mux” | No Mux | No Mux | No Mux | (P1,P2)→(P1,P2) | No Mux | No Mux | No Mux | No Mux | (P1,P2)→(P1,P2) |
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In the offer, ““a=rtcp-mux”” is signaled in the SDP in all the media "m=" lines for which RTP/RTCP mux is required. If the answerer wishes to multiplex RTP and RTCP on a single port, it must generate an answer containing an "“a=rtcp-mux”" attribute for each media where it uses RTP and RTCP-Mux. If the answerer rejects the usage of RTP/RTCP-Mux, it must not associate an SDP "rtcp-mux" or SDP "rtcp" attribute in the answer. Based on the port number, the answerer uses the next higher (odd) destination port number for sending RTCP packets associated with the corresponding "m=" line towards the offerer.RTCP-Mux to non-RTCP-Mux Inter-working for ICE Calls
When using ICE where RTP and RTCP are sent on separate ports, the
performs connectivity checks for each component. Some of these connectivity checks can be avoided by multiplexing RTP and RTCP on the same port, thus reducing the ICE overhead.
When the
offers the “a=rtcp-mux” attribute, it generates the offer that contains "a=candidate:" lines for both RTP and RTCP along with an "a=rtcp:" line (provided that the
sendRtcpPortInSdp
flag is enabled) to designate a fallback port for RTCP in case the answerer does not support RTP and RTCP-Mux. This response is performed for each media stream when RTP and RTCP-Mux is desired.
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With an upgrade to software supporting RTCP-Mux, an existing user running WRTC calls must enable the control rtcpMux on both the PSPs to maintain backward compatibility. This is required when the sessions transparently relay DTLS and RTCP-Mux attributes using DTLS-SRTP relay control (dtlsSrtpRelay ) in the PSP and the flag sdpAttributesSelectiveRelay is enabled. |
Once the RTCP-Mux is negotiated, the subsequent reINVITE(s) sent from the
to the same endpoint does not re-negotiate the RTCP-Mux. However, the RTCP-Mux is re-negotiated with the target endpoints for the redirect calls or for the call transfer. The RTCP-Mux is also negotiated on a per leg basis, when the endpoint triggers the negotiation or the SDP change includes an address change. Thus, the
never restarts ICE on its own and waits for the endpoint to trigger an ICE restart. If a new media stream is added in an existing session, ICE is negotiated on that stream. If the existing streams are updated for codec changes, ICE is not re-negotiated. The RTCP-Mux re-negotiation is consistent for the
both for the ICE and the non-ICE scenarios.
Limitation
- The does not support pass-through for the rtcp-mux.
Asymmetric PRACK Inter-working
The
supports PRACK functionalities either on the ingress or on the egress leg. This enhancement is denoted by the phrase "Asymmetric PRACK Interworking". The flag
sdp100relIwkForPrack
supports Asymmetric PRACK Inter-working for the late media calls. Note |
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To configure PRACK on both the legs, enable the existing flag endToEndPrack . For more information on endToEndPrack , refer to Flags - CLI. |
Anchor |
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| Supported Call Flow Scenarios |
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| Supported Call Flow Scenarios |
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Supported Call Flow Scenarios:
The configurations for the existing flags associated with the flag sdp100relIwkForPrack
, and the supported call flow scenarios, are summarized below:
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0 | Table |
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1 | Parameter configurations for supported call flows |
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rel100Support flag on the ingress Trunk Group | rel100Support flag on the egress Trunk Group | lateMediaSupport flag on the ingress Trunk Group | sdp100relIwkForPrack flag on the egress Trunk Group | Relevant Call Flow Scenario |
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enabled | disabled | enabled | passthru | enabled | Scenario 1 | enabled | disabled | enabled | convert | enabled | Scenario 2 | disabled | enabled | passthru | enabled | Scenario 3 |
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Info |
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For more information on the existing flags lateMediaSupport and rel100Support , refer to the following pages: lateMediaSupport :rel100Support :
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Scenario 1: When PRACK is supported only on the ingress leg of the call, and the
lateMediaSupport
flag is set to
passthru
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0 | Figure |
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1 | Asymmetric PRACK Interworking - Supported Call Flow Scenario 1 |
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Scenario 2: When PRACK is supported only on the ingress leg of the call, and the
lateMediaSupport
flag is set to
convert
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1 | Asymmetric PRACK Interworking - Supported Call Flow Scenario 2 |
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Scenario 3: When PRACK is supported only on the egress leg of the call, and the
lateMediaSupport
flag is set to
passthru
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0 | Figure |
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1 | Asymmetric PRACK Interworking - Supported Call Flow Scenario 3 |
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