In this section:
Ribbon recommends using the Transparency Profile to configure transparency on the SBC Core for new deployments, as well as applying additional transparency configurations to existing deployments. Do not use IP Signaling Profile flags in these scenarios because the flags will be retired in upcoming releases. Refer to the SBC SIP Transparency Implementation Guide for additional information.
SDP Transparency
Session Description Protocol (SDP) is a set of rules defining how to set the multimedia sessions to allow all endpoints to participate in the session effectively.
Interactive Connectivity Establishment (ICE) is a protocol used by systems that cannot determine their 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
SDP Transparency functionality includes:
- Passing all SDP attributes transparently
- Dropping all unknown components of known SDP attributes.
- Dropping any unknown audio codecs.
- Transparently passing all known and unknown video codecs.
When SDP Transparency is enabled, the
For additional audio/video support topics, refer to:
Suppressing 183 Response Without SDP
The
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
suppress183For3xxRedirectResponse
suppress183WithoutSdp
% set profiles signaling ipSignalingProfile <SIP profile name> ingressIpAttributes flags suppress183For3xxRedirectResponsek <disable | enable> % set profiles signaling ipSignalingProfile <SIP profile name> ingressIpAttributes flags suppress183WithoutSdp <disable | enable>
The following table describes the effects of enabling/disabling these flags.
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
The
Late Media INVITE or Re-INVITE Support
Do not use deriveFromOtherLeg
when configuring H.323 or SIP trunk groups to use INVITEs with no SDPs.
sendAllAllowedCodecsForLateMediaInviteOrReInvite
flag controls the handling of audio codecs that the - When this flag is in disabled state for transcoded calls (default behavior), the Unable to show "metadata-from": No such page "_space_variables"offers the codec used for transcoding on the leg.
- When this flag is in enabled state for transcoded calls, the Unable to show "metadata-from": No such page "_space_variables"offers multiple codecs which include:
- The subset of the codecs that the associated peer supports.
- The transcoded codecs that the associated DSP channel supports which includes the codec currently used for transcoding.
For pass-through calls, the
Example 1: G711 pass-through call
- Ingress Offer: G711, G729
- Egress Offer: G711, G729, AMR, G726
- Ingress Answer: G711
- Egress Answer: G711
If the
Example 2: G711 to AMR transcoded call
- Ingress Offer: G711, G729
- Egress Offer: G711,G729, AMR, G726
- Egress Answer: AMR
- Ingress Answer: G711
If theUnable to show "metadata-from": No such page "_space_variables"receives an re-INVITE with no SDP on egress, by default, theUnable to show "metadata-from": No such page "_space_variables"generates an offer for AMR. If the “Send All Allowed Codecs For Late Media Invite Or Re-Invite” flag is in enabled state, then theUnable to show "metadata-from": No such page "_space_variables"generates a codec list of G711,G729, AMR, and G726 in the offer.
The SBC has an existing flag for late media INVITEs or Re-INVITES which specifies the included codecs in the SDP offer, but the behavior differs based on whether the base call is a pass-through or transcoded call. For pass-through calls, the codecs offered do not include those supported for transcoding. For certain scenarios, the originator of the INVITE/Re-INVITE must know the full capabilities even for a pass-through call. This feature addresses the pass-through case where the full set of codecs is needed.
When the sendSBCSupportedCodecsForLateMediaReInvite
flag is enabled, the SBC sends all the codecs supported on a particular leg as an SDP OFFER in the 200 OK response to a late media Re-INVITE. The OFFER sent in the 200 OK contains all the pass-through and transcodable codecs supported on that leg. This behavior is applicable only for late media Re-INVITE requests. The flag does not have any impact on late media INVITE requests.
For example, the
sendSBCSupportedCodecsForLateMediaReInvite
flag is enabled, the sendAllAllowedCodecsForLateMediaInviteOrReInvite
flag and the sendOnlyPreferredCodec
flag in the IP Signaling profile. You must set the SIP trunk group media flag lateMediaSupport
to "convert" to apply the behavior enabled by the sendSBCSupportedCodecsForLateMediaReInvite
flag. Refer to Common IP Attributes - SIP - CLI or Common IP Attributes - Flags (EMA) for configuration details.
Late Media Pass-through Calls over GW-GW
The SBC supports late media pass-through calls over GW-GW and also handles late media (offer-less) re-INVITEs on either side. This functionality is supported when the flag lateMediaSupport
is configured as passthru.
set addressContext <addressContext name> zone <zone name> sipTrunkGroup <sipTrunkGroup name> media lateMediaSupport passthru
When the flag lateMediaSupport
is set as passthru
, a late media INVITE/re-INVITE received on one leg is relayed to the other leg.
The cloud-based SBC SWe does not support Late Media pass-through calls over GW-GW.
ANAT Support
Alternative Network Address Types (ANAT) semantics for the SDP grouping framework allows the expression of alternative network addresses (for example. different IP versions) for a particular media stream. This ability is useful in environments with both IPv4-only hosts and IPv6-only hosts.
The
ICE-Lite Support
Overview
Interactive Connectivity Establishment (ICE) is a protocol for Network Address Translator (NAT) traversal for multimedia sessions established with the offer/answer model. ICE uses the Session Traversal Utilities for NAT (STUN) protocol and its extension Traversal Using Relay NAT (TURN). ICE uses STUN and TURN servers to overcome network address translation issues that can occur when an endpoint is situated behind a NAT device. ICE solves the NAT issues for media streams. Support of the ICE specification is required by WebRTC (WRTC) endpoints.
The SBC is capable of acting as an ICE-Lite agent to allow the WRTC endpoints to connect to the existing VoIP network through the SBC using the DTLS-SRTP protocol. While acting as an ICE-Lite endpoint, the SBC interconnects with endpoints that have either ICE-Lite or full-ICE implementations. ICE-Lite is a lite version of the ICE protocol, which is defined to exchange media with each other. The interconnect with ICE-Lite end points is only for the purpose of IPv4 to IPv6 inter-working and no NAT traversal procedures are supported for ICE-Lite to ICE-Lite.
As SBC is not required to act as an ICE client (it is not located behind a NAT in current deployment models), the initial support for ICE is limited to the SBC acting as an ICE-Lite endpoint.
ICE-Lite key points:
STUN message processing:
- Decoding received messages on media port
- Validating the received messages
- Authenticating the received messages
- Encoding outgoing STUN Response messages and sending on allocated media port
- Encrypting outgoing STUN as per the configured username/password
Reachable IP address: an ICE-Lite agent has a reachable public IP address and can work with other agents that use ICE and are behind the NAT.
ICE-Lite Implementation
The ICE-Lite procedure starts with the offerer discovering all IP ports where it is reachable.These are known as local “candidates”. The “candidates” can be any or all of the following:
- Host candidates (Local IP port)
- Server Reflexive Candidates (External IP port allocated by NAT)
- Relay candidates (IP port on a media relay)
Use STUN to discover the Server Reflexive and Peer reflexive addresses on the NAT. Use TURN to discover the Relay candidates on the TURN relay. After the local candidates are discovered, the offerer (Full ICE Agent) sends them in the session description protocol (SDP) offer to the remote endpoints. The remote end point (SBC) discovers its own local candidates (which are only host candidates) and sends in the answer SDP to the offerer.
Once the offer/answer exchange completes, the Full ICE Agent assumes the role of ICE controlling agent and performs connectivity check between the candidate pairs (from local candidate to remote candidate) by sending the STUN messages. The SBC acts as a controlled agent, and responds to the STUN requests. Based on the success/failure of the connectivity check, the controlling agent selects a candidate pair to exchange the media, and sends a connectivity check on the selected pair with USE-CANDIDATE attribute to let the controlled agent know about the selected pair. The connectivity check may lead to the discovery of new local candidates due to the presence of Restricted or Symmetric NAT. These local candidates are also included in the procedure and are known as “Peer Reflexive Addresses”. The SBC, as a controlled ICE-Lite agent, cuts through the media after receiving USE-CANDIDATE on all components of the media stream (for example, if RTP and RTCP are required for a stream, the media is only cut-through when USE-CANDIDATE is received for both RTP and RTCP).
The controlling agent sends an updated offer by using the selected local candidate in the default IP port of the media line, and the controlled agent (SBC) responds by sending its local candidate in the default IP port of the media line to complete the ICE-Lite procedure. Use the Full ICE agent (using the ICE Restart procedures) to restart the Ice-lite procedure if the Media ports change during a call.
The ICE-Lite procedure starts with the offerer discovering all IP ports where it is reachable
Bundling of Streams
The
sdpAttributesSelectiveRelay
flag 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.
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 using RTP/RTCP multiplexing, use the same address:port combination for all RTP and RTCP packets.
Single-bundle
The
Stream
You can tag a stream as 'bundle-only' with the 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 interworking causes the streams to unbundle.
Bundle Only
The
- 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 Unable to show "metadata-from": No such page "_space_variables"treats the stream as a bundle-only stream.
- The Unable to show "metadata-from": No such page "_space_variables"handles media attributes received with a bundle-only stream offer, taking into consideration the stream is present, as well as the output in the outbound offers (For example, media codecs and fmtp options).
- The handles the media attributes received with a bundle-only stream offer taking into consideration that the stream is present and output in the outbound offers. For example; media codecs, fmtp options)
Bundle Group
The
- The OFFER contains the BUNDLE attribute followed by a list of streams, which are part of the BUNDLE.
- The OFFER contains a unique address:port combinations for each stream as the peer may not support bundling the streams.
If the peer supports 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
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
ANSWER
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
Interactions with Security
Bundling is only allowed when security is relayed, and:
- DTLS-SRTP and DTLS-SCTP for data are relayed
- SDES SRTP is relayed
Limitations
- You can only relay (not interwork) bundling.
- Configuration for LI or a call requiring transcoding, DTMF, or security termination is unbundled.
- Bundling is not supported over the gateway protocol.
RTP and RTCP Multiplexing
The
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
- The Unable to show "metadata-from": No such page "_space_variables"allows relay of multiplexed RTP and RTCP traffic only when it supports the rtcp-mux 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 the Unable to show "metadata-from": No such page "_space_variables"uses rtcp-mux within a bundle, rtcp-mux is enabled for all streams in a bundled or disabled for all.
Interactions with ICE
Using media stream bundling and 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 the RTP.
- A bundle-only "m=" line does not include an ICE candidate or ufrag/pwd.
For the ANSWER:
- The ICE data matches the chosen master stream.
- A single ICE candidate line is included for the RTP component
RTCP Multiplexing to Non-RTCP Multiplexing Interworking
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 considering maintaining multiple NAT bindings is costly.To overcome these scenarios, the
Prior to the 6.1.0 release, the
The
rtcpMux
flag is added to the PSP to configure RTCP-Mux.When the - Multimedia calls
- Monitoring calls
- Both direct media and OMR call scenarios
- DTLS and SRTP scenarios (irrespective of whether the encryption protocol terminates at the Unable to show "metadata-from": No such page "_space_variables"or relays transparently end-to-end)
- Multiplexing RTCP and interworking 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 ( The Unable to show "metadata-from": No such page "_space_variables"supports this in both the cases of SRTP pass-through and SRTP to RTP interworking.)
RTCP bandwidth requirements do not change with the use of RTCP-Mux. The
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 interworking scenarios for offer/answer:
In the offer, ““a=rtcp-mux”” is signaled in the SDP in all the media "m=" lines for whichit requires RTP/RTCP mux. 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 interworking for ICE Calls
When using ICE where the
When 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 it desire the RTP and RTCP-Mux.With an upgrade to
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
RTCP-Mux re-negotiation is consistent for both for the
Limitation
- The Unable to show "metadata-from": No such page "_space_variables"does not support pass-through for the
rtcp-mux.
Asymmetric PRACK interworking
The
sdp100relIwkForPrack
supports Asymmetric PRACK interworking for the late media calls.Supported Call Flow Scenarios:
The configurations for the existing flags associated with the flag sdp100relIwkForPrack
, and the supported call flow scenarios, are summarized below:
Scenario 1: When PRACK is supported only on the ingress leg of the call, and the lateMediaSupport
flag is set to passthru
Scenario 2: When PRACK is supported only on the ingress leg of the call, and the lateMediaSupport
flag is set to convert
Scenario 3: When PRACK is supported only on the egress leg of the call, and the lateMediaSupport
flag is set to passthru
Multiple m-lines Support in SDP
The
When the
Support for multiple m-lines is enabled or disabled using the multipleAudioStreamsSupport
option within media parameters in SIP trunk groups.
If you enable multiple m-line support, you also have the option to enable a separate media option in SIP trunk groups, which disables all SRTP (SAVP) streams present in incoming SDP. This disabling enables deployments to disallow SRTP on the ingress call leg, but not on the egress leg. Enabling the option disallowSrtpStream
means that for all SAVP streams in the incoming SDP, the
disallowSrtpStream
can only be enabled when the multipleAudioStreamsSupport
option is enabled.The multiple m-line support applies only to cases where the m-lines present consist of all audio streams or a combination of audio and image media streams. The
When the multipleAudioStreamsSupport
option is disabled, the SBC default behavior takes place.
For more information, refer to:
Offer-Answer Timer Configuration
In specific call scenarios, the
Currently, the internal Offer-Answer (OA) timer value is fixed and cannot be configured. To overcome this limitation, the offerAnswerTimer
to configure this OA timer.
Call flow
The following figure depicts the call message flow diagram in the Offer-Answer cycle: