Question:
The issuance overview step 3 says that the Client and Attester interact to do whatever is necessary to convince the Attester to attest. But there doesn't seem to be any pretext for this process to be initiated. Step 4 says that the client requests issuance (via the Attester), which would provide the necessary cause to start the attesting process. Should these steps be swapped?
Either the attester is an intermediary to the flow (which is what we're doing in practice), or it is a separate flow. So the prerequisite is that the client knows it wants to request a token from an issuer, and then it does attestation.
Perhaps this could be made more clear if we say that the client, upon deciding it wants to invoke the issuance protocol, then begins the deployment-specific attestation process with its trusted attester? I can suggest some text for this.

If the Client already has a token available that satisfies the token challenge, e.g., because the Client has a cache of previously issued tokens, it can skip to step-redemption and <del> redeem its token.  Otherwise, it invokes the issuance protocol to request a token from the designated Issuer. The first step in the issuance protocol is attestation. Specifically, the Attester performs attestation checks on the Client.  These checks could be proof of solving a CAPTCHA, device trust, hardware attestation, etc (see attester). If attestation succeeds, the client creates a Token Request to send to the designated Issuer (generally via the Attester).  The Attester and Issuer might be functions on the same server, depending on the deployment model (see deployment).  Depending on the details of Attestation, the Client can send the Token Request to the Attester alongside any attestation information.  If attestation fails, the Client receives an error and issuance aborts without a token. </del> <ins> redeem its token. If the Client does not have a token available and decides it wants to obtain one (or more) bound to the token challenge, it then invokes the issuance protocol.  As a prerequisite to the issuance protocol, the Client runs the deployment specific attestation process that is required for the designated Issuer.  Client attestation can be done via proof of solving a CAPTCHA, checking device or hardware attestation validity, etc; see attester for more details. If the attestation process completes successfully, the client creates a Token Request to send to the designated Issuer (generally via the Attester).  The Attester and Issuer might be functions on the same server, depending on the deployment model (see deployment).  Depending on the attestation process, it is possible for attestation to run alongside the issuance protocol, e.g., where Clients send necessary attestation information to the Attester along with their Token Request.  If the attestation process fails, the Client receives an error and issuance aborts without a token. </ins> The Issuer generates a Token Response based on the Token Request, which is returned to the Client (generally via the Attester).


Question:
I think that this definition is incorrect. The latest sentence is from the definition of Origin-Client unlinkability:
good catch. Copy-pasta error.

Issuer-Client unlinkability.  This is similar to Origin-Client unlinkability in that a Client in an issuance context is indistinguishable from any other Client that might use the same <del> issuance context.  The set of Clients that share the same redemption context is referred to as a redemption anonymity set. </del> <ins> issuance context.  The set of Clients that share the same issuance context is referred to as an issuance anonymity set. </ins> Attester-Origin unlinkability.  This is similar to Origin-Client and Issuer-Client unlinkability.  It means that given two attestation contexts, the Attester cannot determine if both contexts correspond to the same Origin or two different Origins. The set of Clients that share the same attestation context is <del> referred to as an anonymity set. </del> <ins> referred to as an attestation anonymity set. </ins> By ensuring that different contexts cannot be linked in this way, only the Client is able to correlate information that might be used


Question:
NAME good suggestions! I merged them all.
This system extensively uses tokens that are only loosely bound to usage contexts. Possession of a token is pretty much all that is necessary. That means that tokens might be stolen or hoarded and then replayed. The architecture draft says nothing about this as a risk, but it probably should.
Agree, especially for for cached tokens. I thought we had mentioned this somewhere, but if not, let's make sure it's included.

If the Client already has a token available that satisfies the token challenge, e.g., because the Client has a cache of previously issued tokens, it can skip to step-redemption and <del> redeem its token. </del> <ins> redeem its token; see hoarding for security considerations of cached tokens. </ins> If the Client does not have a token available and decides it wants to obtain one (or more) bound to the token challenge, it then


Question:
NAME good suggestions! I merged them all.
This system extensively uses tokens that are only loosely bound to usage contexts. Possession of a token is pretty much all that is necessary. That means that tokens might be stolen or hoarded and then replayed. The architecture draft says nothing about this as a risk, but it probably should.
Agree, especially for for cached tokens. I thought we had mentioned this somewhere, but if not, let's make sure it's included.

Origins.  Implementing mitigations discused in deployment and privacy is therefore necessary to ensure that Privacy Pass offers meaningful privacy improvements to end-users. <ins> 7.1. Depending on the Origin's token challenge, Clients can request and cache more than one token using an issuance protocol.  Cached tokens help improve privacy by separating the time of token issuance from the time of token redemption, and also allow Clients to reduce the overhead of receiving new tokens via the issuance protocol. As a consequence, Origins that send token challenges which are compatible with cached tokens need to take precautions to ensure that tokens are not replayed.  This is typically done via keeping track of tokens that are redeemed for the period of time in which cached tokens would be accepted for particular challenges. Moreover, since tokens are not intrinsically bound to Clients, it is possible for malicious Clients to collude and share tokens in a so- called "hoarding attack."  As an example of this attack, many distributed Clients could obtain cacheable tokens and them share them with a single Client to redeem in a way that would violate an Origin's attempt to limit tokens to any one particular Client. Depending on the deployment model, it can be possible to detect these types of attacks by comparing issuance and redemption contexts; for example, this is possible in the Joint Origin and Issuer model. </ins>


Question:
In a successful issuance exchange, the attester provides the issuer with information about the client. The text in Section 3.2.1 points out that the way in which information about the client is combined could reduce anonymity set size in ways that affect privacy. Given that we have a chain of custody, I imagine that the privacy property here is about what the issuer learns about a client. More information means more information that might then propagate from the issue onward to the origin. This means that the attester needs to limit what it passes along to only that which is relevant. The client needs to trust the attester to do this. ... or does it? The text in Section 3.2.1 implies strongly that this anonymity set size is important, but the constraints on the client-issuer interaction, especially consistency rules, very much limits what the issuer can pass along to the origin. So maybe the system depends less on trusting the attester and more on the combination of the cryptographic properties of the token and the consistency system that mitigates against misuse. None of this appears in the privacy considerations. Should it?
On this point: "Clients do not explicitly trust Issuers." (Section 3.2.2) This would invalidate much of the discussion of anonymity sets in attestations. This is quite confusing. The client doesn't need to trust the issuer, because the issuer is unable to pass information on to others. However, the client cannot trust the issuer with information, so it has to trust the attester not to pass stuff along AND it has to use an anonymizing proxy to interact with the issuer so that nothing leaks. This is quite contradictory overall.
There are issuance protocols wherein the issuer learns information about the origin by design, like the . In this variant, if the attester were to pass information about the client on to the issuer, then the issuer would be able to reconstruct the (client, origin) pair, breaking the unlinkability goals. I agree that in the case where the issuer learns nothing about the origin in the protocol then reliance on the attester is lessened, but we need to make sure we accommodate future issuance protocols here. I'll think about ways in which we can try to make this more clear. This might go into the privacy considerations, as suggested.

3.4.1. <del> Attestation is an important part of the issuance protocol.  In Privacy Pass, attestation is the process by which an Attester bears witness to, confirms, or authenticates a Client so as to verify a property about the Client that is required for Issuance.  Clients explicitly trust Attesters to perform attestation correctly and in a way that does not violate their privacy. </del> <ins> In Privacy Pass, attestation is the process by which an Attester bears witness to, confirms, or authenticates a Client so as to verify properties about the Client that are required for Issuance.  Issuers trust Attesters to perform attestation correctly. </ins> RFC9334 describes an architecture for attestation procedures.  Using that architecture as a conceptual basis, Clients are RATS attesters


Question:
In a successful issuance exchange, the attester provides the issuer with information about the client. The text in Section 3.2.1 points out that the way in which information about the client is combined could reduce anonymity set size in ways that affect privacy. Given that we have a chain of custody, I imagine that the privacy property here is about what the issuer learns about a client. More information means more information that might then propagate from the issue onward to the origin. This means that the attester needs to limit what it passes along to only that which is relevant. The client needs to trust the attester to do this. ... or does it? The text in Section 3.2.1 implies strongly that this anonymity set size is important, but the constraints on the client-issuer interaction, especially consistency rules, very much limits what the issuer can pass along to the origin. So maybe the system depends less on trusting the attester and more on the combination of the cryptographic properties of the token and the consistency system that mitigates against misuse. None of this appears in the privacy considerations. Should it?
On this point: "Clients do not explicitly trust Issuers." (Section 3.2.2) This would invalidate much of the discussion of anonymity sets in attestations. This is quite confusing. The client doesn't need to trust the issuer, because the issuer is unable to pass information on to others. However, the client cannot trust the issuer with information, so it has to trust the attester not to pass stuff along AND it has to use an anonymizing proxy to interact with the issuer so that nothing leaks. This is quite contradictory overall.
There are issuance protocols wherein the issuer learns information about the origin by design, like the . In this variant, if the attester were to pass information about the client on to the issuer, then the issuer would be able to reconstruct the (client, origin) pair, breaking the unlinkability goals. I agree that in the case where the issuer learns nothing about the origin in the protocol then reliance on the attester is lessened, but we need to make sure we accommodate future issuance protocols here. I'll think about ways in which we can try to make this more clear. This might go into the privacy considerations, as suggested.

minimizing the set of supported attestation procedures helps minimize the amount of information leaked through a token. <del> Each attestation procedure also has an impact on the overall system privacy.  Requiring a conjunction of attestation types could decrease the overall anonymity set size.  For example, the number of Clients that have solved a CAPTCHA in the past day, that have a valid account, and that are running on a trusted device is less than the number of Clients that have solved a CAPTCHA in the past day. Attesters SHOULD not admit attestation types that result in small anonymity sets. </del> <ins> The role of the Attester in the issuance protocol and its impact on privacy depends on the type of attestation procedure, issuance protocol, deployment model.  For instance, requiring a conjunction of attestation procedures could decrease the overall anonymity set size. As an example, the number of Clients that have solved a CAPTCHA in the past day, that have a valid account, and that are running on a trusted device is less than the number of Clients that have solved a CAPTCHA in the past day.  Attesters SHOULD NOT be based on attestation procedures that result in small anonymity sets. Depending on the issuance protocol, the Issuer might learn information about the Origin.  To ensure Issuer-Client unlinkability, the Issuer should be unable to link that information to a specific Client.  For such issuance protocols where the Attester has access to Client-specific information, such as is the case for attestation procedures that involve Client-specific information (such as application-layer account information) or for deployment models where the Attester learns Client-specific information (such as Client IP addresses), Clients trust the Attester to not share any Client- specific information with the Issuer.  In deployments where the Attester does not learn Client-specific information, the Client does not need to explicitly trust the Attester in this regard. </ins> Issuers trust Attesters to correctly and reliably perform attestation.  However, certain types of attestation can vary in value


Question:
8445 8174 Future documents, not future versions Fix normative refs
Sure - happy to add text of this sort if we think it would be valuable. OK. :-) OK.
LGTM

WebRTC IP Address Handling Requirements <del> draft-ietf-rtcweb-ip-handling-10 </del> <ins> draft-ietf-rtcweb-ip-handling-11 </ins> Abstract


Question:
8445 8174 Future documents, not future versions Fix normative refs
Sure - happy to add text of this sort if we think it would be valuable. OK. :-) OK.
LGTM

2. The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", <del> "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC2119. </del> <ins> "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 RFC2119 RFC8174 when, and only when, they appear in all capitals, as shown here. </ins> 3. In order to establish a peer-to-peer connection, WebRTC implementations use Interactive Connectivity Establishment (ICE) <del> RFC5245, which attempts to discover multiple IP addresses using </del> <ins> RFC8445, which attempts to discover multiple IP addresses using </ins> techniques such as Session Traversal Utilities for NAT (STUN) RFC5389 and Traversal Using Relays around NAT (TURN) RFC5766, and then checks the connectivity of each local-address-remote-address pair in order


Question:
8445 8174 Future documents, not future versions Fix normative refs
Sure - happy to add text of this sort if we think it would be valuable. OK. :-) OK.
LGTM

choose stricter modes if desired, e.g., if a user indicates they want all WebRTC traffic to follow the default route. <del> Future versions of this document may define additional modes and/or update the recommended default modes. </del> <ins> Future documents may define additional modes and/or update the recommended default modes. </ins> Note that the suggested defaults can still be used even for organizations that want all external WebRTC traffic to traverse a


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

1.3. <del> When RFC8829 was published, inconsistencies regarding BUNDLE RFC8843 operation were identified with regard to both the specification text as well as implementation behavior.  The former concern was addressed via an update to RFC8843.  For the latter concern, it was observed that some implementations implemented the "max-bundle" bundle policy defined in RFC8829 by assuming that bundling had already been negotiated, rather than marking "m=" sections as bundle-only as indicated by the specification.  In order to prevent unexpected changes to applications relying on the pre-standard behavior, the decision was made to deprecate "max-bundle" and instead introduce an identically defined "must-bundle" policy that, when selected, provides the behavior originally specified by RFC8829. </del> <ins> When RFC8829 was published, inconsistencies regarding BUNDLE I- D.ietf-mmusic-rfc8843-bis operation were identified with regard to both the specification text as well as implementation behavior.  The former concern was addressed via an update to I-D.ietf-mmusic- rfc8843-bis.  For the latter concern, it was observed that some implementations implemented the "max-bundle" bundle policy defined in RFC8829 by assuming that bundling had already been negotiated, rather than marking "m=" sections as bundle-only as indicated by the specification.  In order to prevent unexpected changes to applications relying on the pre-standard behavior, the decision was made to deprecate "max-bundle" and instead introduce an identically defined "must-bundle" policy that, when selected, provides the behavior originally specified by RFC8829. </ins> 2.


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

bundled (either by a successful bundle negotiation or by being marked as bundle-only), then candidates will be gathered and exchanged for that "m=" section if and only if its MID item is a BUNDLE-tag, as <del> described in RFC8843. </del> <ins> described in I-D.ietf-mmusic-rfc8843-bis. </ins> 3.5.2.


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

MUST be zero. The application can specify its preferred policy regarding the use of <del> BUNDLE, the multiplexing mechanism defined in RFC8843.  Regardless of policy, the application will always try to negotiate bundle onto a single transport and will offer a single bundle group across all "m=" sections; use of this single transport is contingent upon the answerer accepting bundle.  However, by specifying a policy from the list below, the application can control exactly how aggressively it will try to bundle media streams together, which affects how it will interoperate with a non-bundle-aware endpoint.  When negotiating with a non-bundle-aware endpoint, only the streams not marked as bundle- only streams will be established. </del> <ins> BUNDLE, the multiplexing mechanism defined in I-D.ietf-mmusic- rfc8843-bis.  Regardless of policy, the application will always try to negotiate bundle onto a single transport and will offer a single bundle group across all "m=" sections; use of this single transport is contingent upon the answerer accepting bundle.  However, by specifying a policy from the list below, the application can control exactly how aggressively it will try to bundle media streams together, which affects how it will interoperate with a non-bundle- aware endpoint.  When negotiating with a non-bundle-aware endpoint, only the streams not marked as bundle-only streams will be established. </ins> The set of available policies is as follows:


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

RFC8859 groups SDP attributes into different categories.  To avoid unnecessary duplication when bundling, attributes of category IDENTICAL or TRANSPORT MUST NOT be repeated in bundled "m=" sections, <del> repeating the guidance from RFC8843.  This includes "m=" sections for which bundling has been negotiated and is still desired, as well as "m=" sections marked as bundle-only. </del> <ins> repeating the guidance from I-D.ietf-mmusic-rfc8843-bis.  This includes "m=" sections for which bundling has been negotiated and is still desired, as well as "m=" sections marked as bundle-only. </ins> The following attributes, which are of a category other than IDENTICAL or TRANSPORT, MUST be included in each "m=" section:


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

possible, and media sections will not be marked as bundle-only.  This is by design, but could cause issues in the rare case of sending a subsequent offer as an initial offer to a non-bundle-aware endpoint <del> via Third Party Call Control (3PCC). </del> <ins> via Third Party Call Control (3PCC), as discussed in I-D.ietf-mmusic- rfc8843-bis. </ins> "a=group:LS" attributes are generated in the same way as for initial offers, with the additional stipulation that any lip sync groups that


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

If the answer contains valid bundle groups, discard any ICE components for the "m=" sections that will be bundled onto the primary ICE components in each bundle, and begin muxing these "m=" <del> sections accordingly, as described in RFC8843. </del> <ins> sections accordingly, as described in I-D.ietf-mmusic-rfc8843-bis. </ins> If the description is of type "answer" and there are still remaining candidates in the ICE candidate pool, discard them.


Question:
and
To resolve WGLC comments: URL it was agreed to incorporate a reference to this text in 8843bis: URL That text ended up in s7.6.
When draft-8843bis is given an RFC number, this draft should reference the new RFC. Hi, Eventhough I would not like to make more changes than necessary, I am fine with "3PCC Considerations". However, your suggested text is very difficult to understand in some places, so let me give it a try. (The first paragraph is generic, the second SIP specific, and the third BUNDLE specific.) 3PCC Considerations In some 3PCC scenarios a new session will be established between an endpoint that is currently part of an ongoing session and an endpoint that is currently not part of an ongoing session. The endpoint that is part of a session will generate a subsequent offer that will be forwarded to the other endpoint by a 3PCC controller. The endpoint that is not part of a session will process the offer as an initial offer. The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If the UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA). If the UA is not part of an ongoing session, it will process the offer as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers, and it cannot be assumed that a UA is able to correctly process a subsequent offer as an initial offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer, following the procedures in (Section 7.2), before it forwards the offer to a UA. In the rewritten offer the 3PCC controller will set the port value to zero (and include an SDP 'bundle-only' attribute) for each "m=" section within the BUNDLE group, excluding the offerer-tagged "m=" section. __ From: Roman Shpount Sent: Tuesday, November 2, 2021 6:33 PM To: Christer Holmberg Cc: Justin Uberti ; Justin Uberti ; RTCWeb IETF Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL How about we replace the SIP Considerations with: 3PCC Considerations In some 3PCC scenarios, an offer generated during an ongoing session, i.e., a subsequent offer, will be used by a 3PCC controller to establish a new session and forwarded as an initial offer to another endpoint that is currently not part of a session. For example, the Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. If UAS is a part of an ongoing session, it will include a subsequent offer in the 200 OK response. The offer will be received by a 3PCC controller (UAC) and then forwarded to another User Agent (UA) as an initial offer. When the BUNDLE mechanism is used, an initial BUNDLE offer is constructed using different rules than subsequent BUNDLE offers. It cannot be assumed that a subsequent offer is a valid initial offer and that the endpoint that expects an initial offer will properly process such a subsequent offer. Therefore, the 3PCC controller SHOULD rewrite the subsequent offer into a valid initial offer before it is used to establish a new session. To make the subsequent offer a valid initial offer, 3PCC will need to modify all the non-tagged m= lines to include the bundle-only attribute and set the m= line port to zero. _ Roman Shpount Christer Holmberg > wrote: Hi, What about something like this: OLD: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP Offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. >From a BUNDLE perspective, such SDP Offer SHOULD be generated using the procedures defined in Section 7.2.” NEW: “The Session Initiation Protocol (SIP) [RFC3261] allows a User Agent Client (UAC) to send a re-INVITE request without an SDP body (sometimes referred to as an empty re-INVITE). In such cases, the User Agent Server (UAS) will include an SDP offer in the associated 200 OK response. This is typically used for 3rd Party Call Control (3PCC) scenarios. In some 3PCC scenarios the UAS will be part of an ongoing session, and will therefore include a subsequent offer in the 200 OK responses. The offer will be received by a 3PCC controller (UAC) and then forwarded as an initial offer to another User Agent (UA) that is currently not part of a session. When the BUNDLE mechanism is used, as an initial BUNDLE offer look different than a subsequent BUNDLE offer, it cannot be assumed that a UA that expects an initial offer will be able to properly process a subsequent offer. Therefore, the 3PCC controller needs to act as a Back-To-Back User Agent (B2BUA), and when it receives the subsequent offer it needs to rewrite it into an initial offer before it is forwarded to such UA.” Regards, Christer From: Roman Shpount > Sent: tiistai 2. marraskuuta 2021 10.41 To: Justin Uberti > Cc: Christer Holmberg >; Justin Uberti >; RTCWeb IETF > Subject: Re: [rtcweb] Working Group Last Call for draft-uberti-rtcweb-rfc8829bis-URL The PROBLEM is that we have two endpoints, where one sends a subsequent offer, and the other one expects an initial offer. What do you normally do when you have that kind of problem? You use an SBC/B2BUA. In this case that SBC/B2BUA would be the 3PCC controller. So, my suggestion would be to remove the SHOULD text from 8843bis, and simply add a note somewhere (in 8843bis and/or 8829bis) which describes the issue and says that the 3GPP controller needs to modify the offer accordingly. Roman, thoughts on this? If the 3PCC is going to rewrite the offer SDP anyway then maybe adding a=bundle-only isn't the end of the world. I am not opposed to this idea. 3PCC typically knows that the subsequent offer is going to be used as initial, and should be able to rewrite the offer to make it valid. We can change SIP Considerations section in 8843bis (URL), remove the SHOULD, and specify that 3PCC controller should fix the offer. We can then reference this note from 8829bis or restate the same guidance. Roman Shpount
Not seeing any objections for a month then once -02 is submitted we will progress this towards our AD. spt

6. When bundling, associating incoming RTP/RTCP with the proper "m=" <del> section is defined in RFC8843.  When not bundling, the proper "m=" section is clear from the ICE component over which the RTP/RTCP is received. </del> <ins> section is defined in I-D.ietf-mmusic-rfc8843-bis.  When not bundling, the proper "m=" section is clear from the ICE component over which the RTP/RTCP is received. </ins> Once the proper "m=" section or sections are known, RTP/RTCP is delivered to the RtpTransceiver(s) associated with the "m="


Question:
URL of draft-nandakumar. This is a merge of a previously approved PR
lgtm
LGTM
As per discussion in mmusic, to make it abundantly clear that DTLS is to be used even when running over ICE-TCP, the profile name should be TCP/DTLS/RTP/SAVPF. We'll leave the UDP/TLS/RTP/SAVPF one alone, since that ship has sailed, and it's less ambiguous.
Need Suhas to update URL to reflect the above.
For SCTP, URL has the correct UDP/DTLS/SCTP and TCP/DTLS/SCTP things that we want to refer to.
The action item here is to update this document to use the correct fields as agreed above as if draft-nandakumar was done, and then when that document is fixed we will be good.

RFC4566 is the base SDP specification and MUST be implemented. RFC5764 MUST be supported for signaling the UDP/TLS/RTP/SAVPF <del> RFC5764 and TCP/TLS/RTP/SAVPF I-D.nandakumar-mmusic-proto-iana- </del> <ins> RFC5764 and TCP/DTLS/RTP/SAVPF I-D.nandakumar-mmusic-proto-iana- </ins> registration RTP profiles. RFC5245 MUST be implemented for signaling the ICE credentials and


Question:
URL of draft-nandakumar. This is a merge of a previously approved PR
lgtm
LGTM
As per discussion in mmusic, to make it abundantly clear that DTLS is to be used even when running over ICE-TCP, the profile name should be TCP/DTLS/RTP/SAVPF. We'll leave the UDP/TLS/RTP/SAVPF one alone, since that ship has sailed, and it's less ambiguous.
Need Suhas to update URL to reflect the above.
For SCTP, URL has the correct UDP/DTLS/SCTP and TCP/DTLS/SCTP things that we want to refer to.
The action item here is to update this document to use the correct fields as agreed above as if draft-nandakumar was done, and then when that document is fixed we will be good.

5.1.3. For media m= sections, JSEP endpoints MUST support both the "UDP/TLS/ <del> RTP/SAVPF" and "TCP/TLS/RTP/SAVPF" profiles and MUST indicate one of </del> <ins> RTP/SAVPF" and "TCP/DTLS/RTP/SAVPF" profiles and MUST indicate one of </ins> these two profiles for each media m= line they produce in an offer. <del> For data m= sections, JSEP endpoints must support both the "UDP/TLS/ SCTP" and "TCP/TLS/SCTP" profiles and MUST indicate one of these two </del> <ins> For data m= sections, JSEP endpoints must support both the "UDP/DTLS/ SCTP" and "TCP/DTLS/SCTP" profiles and MUST indicate one of these two </ins> profiles for each data m= line they produce in an offer.  Because ICE can select either TCP or UDP transport depending on network conditions, both advertisements are consistent with ICE eventually


Question:
URL of draft-nandakumar. This is a merge of a previously approved PR
lgtm
LGTM
As per discussion in mmusic, to make it abundantly clear that DTLS is to be used even when running over ICE-TCP, the profile name should be TCP/DTLS/RTP/SAVPF. We'll leave the UDP/TLS/RTP/SAVPF one alone, since that ship has sailed, and it's less ambiguous.
Need Suhas to update URL to reflect the above.
For SCTP, URL has the correct UDP/DTLS/SCTP and TCP/DTLS/SCTP things that we want to refer to.
The action item here is to update this document to use the correct fields as agreed above as if draft-nandakumar was done, and then when that document is fixed we will be good.

compatible mode and use AVP timing, i.e., "trr-int=4". For data m= sections, JSEP endpoints MUST support receiving the <del> "UDP/ TLS/SCTP", "TCP/TLS/SCTP", or "DTLS/SCTP" (for backwards </del> <ins> "UDP/ DTLS/SCTP", "TCP/DTLS/SCTP", or "DTLS/SCTP" (for backwards </ins> compatibility) profiles. Note that re-offers by JSEP endpoints MUST use the correct profile


Question:
URL of draft-nandakumar. This is a merge of a previously approved PR
lgtm
LGTM
As per discussion in mmusic, to make it abundantly clear that DTLS is to be used even when running over ICE-TCP, the profile name should be TCP/DTLS/RTP/SAVPF. We'll leave the UDP/TLS/RTP/SAVPF one alone, since that ship has sailed, and it's less ambiguous.
Need Suhas to update URL to reflect the above.
For SCTP, URL has the correct UDP/DTLS/SCTP and TCP/DTLS/SCTP things that we want to refer to.
The action item here is to update this document to use the correct fields as agreed above as if draft-nandakumar was done, and then when that document is fixed we will be good.

To properly indicate use of DTLS, the <proto> field MUST be set to "UDP/TLS/RTP/SAVPF", as specified in RFC5764, Section 8, if the <del> default candidate uses UDP transport, or "TCP/TLS/RTP/SAVPF", as </del> <ins> default candidate uses UDP transport, or "TCP/DTLS/RTP/SAVPF", as </ins> specified inI-D.nandakumar-mmusic-proto-iana-registration if the default candidate uses TCP transport.


Question:
URL of draft-nandakumar. This is a merge of a previously approved PR
lgtm
LGTM
As per discussion in mmusic, to make it abundantly clear that DTLS is to be used even when running over ICE-TCP, the profile name should be TCP/DTLS/RTP/SAVPF. We'll leave the UDP/TLS/RTP/SAVPF one alone, since that ship has sailed, and it's less ambiguous.
Need Suhas to update URL to reflect the above.
For SCTP, URL has the correct UDP/DTLS/SCTP and TCP/DTLS/SCTP things that we want to refer to.
The action item here is to update this document to use the correct fields as agreed above as if draft-nandakumar was done, and then when that document is fixed we will be good.

Lastly, if a data channel has been created, a m= section MUST be generated for data.  The <media> field MUST be set to "application" <del> and the <proto> field MUST be set to "UDP/TLS/SCTP" if the default candidate uses UDP transport, or "TCP/TLS/SCTP" if the default </del> <ins> and the <proto> field MUST be set to "UDP/DTLS/SCTP" if the default candidate uses UDP transport, or "TCP/DTLS/SCTP" if the default </ins> candidate uses TCP transport I-D.ietf-mmusic-sctp-sdp.  The "fmt" value MUST be set to the SCTP port number, as specified in Section 4.1.  [TODO: update this to use a=sctp-port, as indicated in


Question:
LGTM
http://rtcweb-URL mentions its use at media level; this should be clarified.

Specific processing MUST also be applied for the following attribute lines: <del> If present, a single "a=ice-lite" line is parsed as specified in RFC5245, Section 15.3, and a value indicating the presence of ice- lite is stored. </del> If present, a single "a=ice-ufrag" line is parsed as specified in RFC5245, Section 15.4, and the ufrag value is stored.


Question:
LGTM
Overall looks good - added some minor suggestions
lgtm
F. Section 4.1.1: Regardless of policy, the application will always try to negotiate BUNDLE onto a single transport, and will offer a single BUNDLE group across all media sections. I think this sentence may be a bit misleading without additional sentence(s) to explain that the actual number of established transport streams will depend on the peer's response to the attempt to bundle.
OK. This could be resolved by adding a final sentence of the form "Of course, the answerer can choose whether or not to accept BUNDLE".

BUNDLE, the multiplexing mechanism defined in I-D.ietf-mmusic-sdp- bundle-negotiation.  Regardless of policy, the application will always try to negotiate BUNDLE onto a single transport, and will <del> offer a single BUNDLE group across all media sections.  However, by specifying a policy from the list below, the application can control how aggressively it will try to BUNDLE media streams together, which affects how it will interoperate with a non-BUNDLE-aware endpoint. When negotiating with a non-BUNDLE-aware endpoint, only the streams not marked as bundle-only streams will be established.  The set of available policies is as follows: </del> <ins> offer a single BUNDLE group across all media section; use of this single transport is contingent upon the answerer accepting BUNDLE. However, by specifying a policy from the list below, the application can control exactly how aggressively it will try to BUNDLE media streams together, which affects how it will interoperate with a non- BUNDLE-aware endpoint.  When negotiating with a non-BUNDLE-aware endpoint, only the streams not marked as bundle-only streams will be established.  The set of available policies is as follows: </ins> The first media section of each type (audio, video, or application) will contain transport parameters, which will allow


Question:
b=TIAS is specified in bits per second, b=AS in kbps. Hence the formula needs to multiply by 1000 to change units. Thanks NAME for catching this in code review.
Even though I noticed it in the code review I had not noticed it as an issue in the spec. Good catch!
nice catch - lgtm with minor comment
LGTM
LGTM to me as well. I updated the text in

present, but an "AS" value is specified, generate a "TIAS" value using this formula: <del> TIAS = AS * 0.95 - 50 * 40 * 8 </del> <ins> TIAS = AS * 1000 * 0.95 - 50 * 40 * 8 </ins> The 50 is based on 50 packets per second, the 40 is based on an <del> estimate of total header size, and the 0.95 is to allocate 5% to RTCP.  If more accurate control of bandwidth is needed, "TIAS" should be used instead of "AS". </del> <ins> estimate of total header size, the 1000 is to change the unit from kbps to bps and the 0.95 is to allocate 5% to RTCP.  If more accurate control of bandwidth is needed, "TIAS" should be used instead of "AS". </ins> For any "RR" or "RS" bandwidth values, handle as specified in RFC3556, Section 2.


Question:
LGTM
WFM. NAME PTAL
The current draft suggests moving the discussion of setRemoteDescription: [[OPEN ISSUE: move discussion of setRemoteDescription to the subsequent-offer section.]]

The next step is to generate m= sections, as specified in RFC4566 Section 5.14.  An m= section is generated for each RtpTransceiver <del> that has been added to the PeerConnection via the addTrack, addTransceiver, and setRemoteDescription methods.  [[OPEN ISSUE: move discussion of setRemoteDescription to the subsequent-offer section.]] This is done in the order that their associated RtpTransceivers were added to the PeerConnection and excludes RtpTranscievers that are stopped and not associated with an m= section (either due to an m= section being recycled or an RtpTransceiver having been stopped before being associated with an m= section) . </del> <ins> that has been added to the PeerConnection.  This is done in the order that their associated RtpTransceivers were added to the PeerConnection and excludes RtpTransceivers that are stopped and not associated with an m= section (either due to an m= section being recycled or an RtpTransceiver having been stopped before being associated with an m= section) . </ins> Each m= section, provided it is not marked as bundle-only, MUST generate a unique set of ICE credentials and gather its own unique


Question:
NAME removed the dangerous text.
Stil LGTM
Consider the following sequence of actions: setRemoteDescription(offerwithicerestart); addIceCandidate(candidate); setLocalDescription(answerwithicerestart); As issue describes, when the remote peer has set an offer with an ICE restart, but has not yet set an answer, it's doing ICE processing for both the new and old "generations" simultaneously (assuming we go that route). So, is this a candidate for the new or old generation? If "old", then it can be applied immediately. Otherwise, it should be queued and only applied after setting the local description, since the local description is what triggers the ICE restart. Of course, there's a separate issue, which is that there's no way of telling which "generation" the candidate applies to. One proposed way of fixing this is to trickle a ufrag/password along with the candidate. Another way would be to queue the candidates from the "old generation" before signaling them, and only signal them after rolling back the local description. If instead of rolling back the local description, an answer is applied, these queued old candidates would be discarded. This way, once a candidate is signaled, it unambiguously applies to only the newest generation. Note that the "candidate" could also be an end-of-candidates indication, and the same reasoning applies.
Honghai already has considered how to associate candidates with ufrag/pwds. However, I think in the absence of such markings, the candidate should be associated with the new generation, since the offer specified ICE restart and the candidate followed the offer.
I agree. Thus, if an endpoint gathers a candidate for an old generation (and doesn't have a mechanism to associate it with a ufrag/pwd), it should only signal the candidate if and when the local description that triggered the ICE restart gets rolled back. That's what I was trying to explain in the second-to-last paragraph.
In ICE, we agreed to have a ufrag parameter on the candidate. Where that parameter goes is still TBD; it could be in a=candidate or on RTCIceCandidate. But this makes it clear what the JSEP implementation should do in this case.
The ufrag parameter would need to apply to a=end-of-candidates as well.
This is good news. But now we need to go back to the drawing board with trickling end-of-candidates. If the ufrag needs to be trickled, we can't just trickle a "null" candidate. And we can't trickle a single special candidate with a ufrag, because different media descriptions could have different ufrags.
:-/ Perhaps we could generate individual end-of-candidates objects, followed by a null for backcompat. The null would serve as an implicit end-of-candidates for the current ufrag, but new apps would never use the null.
I think I lean towards having the trickle candidates strongly tied to a ufrag/pwd.
To provide an update on this issue: I think we still want to clarify that if there's a pending ICE restart initiated by the remote peer, any remote candidates from the new "ICE negotiation session" should be queued and only processed by the local ICE agent once the stable signaling state is reached. I don't think this can go in the trickle ICE spec, because trickle ICE has no concept of a pending ICE restart (it's a consequence of having the ability to roll back an ICE restart offer). Also, last I heard the decision was to trickle the ufrag and mid along with end-of-candidates indications. So the second paragraph of the section will need to be changed to reflect this; it's no longer one indication "for all media descriptions in the last remote description".
I've dealt with the second paragraph in PR . The first seems mostly like a "can't happen" to me. As an offerer, you should not be receiving candidates before the answer (that indicates a signaling channel reordering issue). I suppose in principle as an answerer you could receive ice candidates while thinking about the offer, but I think that the answer there is you just discard them.
See also

as defined in RFC5245, Section 4.1.1, unless it has been marked as bundle-only. <del> Or, if the ICE ufrag and password values have changed, trigger the ICE Agent to start an ICE restart and begin gathering new candidates for the media section, as defined in RFC5245, Section 9.1.1.1, unless it has been marked as bundle-only. </del> <ins> Or, if the ICE ufrag and password values have changed, and it has not been marked as bundle-only, trigger the ICE Agent to start an ICE restart, and begin gathering new candidates for the media section as described in RFC5245, Section 9.1.1.1.  If this description is an answer, also start checks on that media section as defined in RFC5245, Section 9.3.1.1. </ins> If the media section proto value indicates use of RTP:


Question:
NAME removed the dangerous text.
Stil LGTM
Consider the following sequence of actions: setRemoteDescription(offerwithicerestart); addIceCandidate(candidate); setLocalDescription(answerwithicerestart); As issue describes, when the remote peer has set an offer with an ICE restart, but has not yet set an answer, it's doing ICE processing for both the new and old "generations" simultaneously (assuming we go that route). So, is this a candidate for the new or old generation? If "old", then it can be applied immediately. Otherwise, it should be queued and only applied after setting the local description, since the local description is what triggers the ICE restart. Of course, there's a separate issue, which is that there's no way of telling which "generation" the candidate applies to. One proposed way of fixing this is to trickle a ufrag/password along with the candidate. Another way would be to queue the candidates from the "old generation" before signaling them, and only signal them after rolling back the local description. If instead of rolling back the local description, an answer is applied, these queued old candidates would be discarded. This way, once a candidate is signaled, it unambiguously applies to only the newest generation. Note that the "candidate" could also be an end-of-candidates indication, and the same reasoning applies.
Honghai already has considered how to associate candidates with ufrag/pwds. However, I think in the absence of such markings, the candidate should be associated with the new generation, since the offer specified ICE restart and the candidate followed the offer.
I agree. Thus, if an endpoint gathers a candidate for an old generation (and doesn't have a mechanism to associate it with a ufrag/pwd), it should only signal the candidate if and when the local description that triggered the ICE restart gets rolled back. That's what I was trying to explain in the second-to-last paragraph.
In ICE, we agreed to have a ufrag parameter on the candidate. Where that parameter goes is still TBD; it could be in a=candidate or on RTCIceCandidate. But this makes it clear what the JSEP implementation should do in this case.
The ufrag parameter would need to apply to a=end-of-candidates as well.
This is good news. But now we need to go back to the drawing board with trickling end-of-candidates. If the ufrag needs to be trickled, we can't just trickle a "null" candidate. And we can't trickle a single special candidate with a ufrag, because different media descriptions could have different ufrags.
:-/ Perhaps we could generate individual end-of-candidates objects, followed by a null for backcompat. The null would serve as an implicit end-of-candidates for the current ufrag, but new apps would never use the null.
I think I lean towards having the trickle candidates strongly tied to a ufrag/pwd.
To provide an update on this issue: I think we still want to clarify that if there's a pending ICE restart initiated by the remote peer, any remote candidates from the new "ICE negotiation session" should be queued and only processed by the local ICE agent once the stable signaling state is reached. I don't think this can go in the trickle ICE spec, because trickle ICE has no concept of a pending ICE restart (it's a consequence of having the ability to roll back an ICE restart offer). Also, last I heard the decision was to trickle the ufrag and mid along with end-of-candidates indications. So the second paragraph of the section will need to be changed to reflect this; it's no longer one indication "for all media descriptions in the last remote description".
I've dealt with the second paragraph in PR . The first seems mostly like a "can't happen" to me. As an offerer, you should not be receiving candidates before the answer (that indicates a signaling channel reordering issue). I suppose in principle as an answerer you could receive ice candidates while thinking about the offer, but I think that the answer there is you just discard them.
See also

parameters are out of bounds, or cannot be applied, processing MUST stop and an error MUST be returned. <del> If the description is of type "offer", and the ICE ufrag or password changed from the previous remote description, as described in Section 9.1.1.1 of RFC5245, mark that an ICE restart is needed. </del> <ins> If the ICE ufrag or password changed from the previous remote description, then an ICE restart is needed, as described in Section 9.1.1.1 of RFC5245 If the description is of type "offer", mark that an ICE restart is needed.  If the description is of type "answer" and the current local description is also an ICE restart, then signal the ICE agent to begin checks as described in Section 9.3.1.1 of RFC5245.  An answer MUST change the ufrag and password in an answer if and only if ICE is restarting, as described in Section 9.2.1.1 of RFC5245. </ins> Configure the ICE components associated with this media section to use the supplied ICE remote ufrag and password for their


Question:
NAME
NAME PTAL
LG with nits
para break and typo comment were missed
In the current draft we have addTransceiver [TODO] We need this text before WGLC, so it's being added to the punchlist.

4.1.3. <del> [TODO] </del> <ins> The addTransceiver method adds a new RTPTransceiver to the PeerConnection.  If a MediaStreamTrack argument is provided, then the transceiver will be configured with that media type and the track will be attached to the transceiver.  Otherwise, the application MUST explicitly specify the type; this mode is useful for creating recvonly transceivers as well as for creating transceivers to which a track can be attached at some later point.  At the time of creation, the application can also specify a transceiver direction attribute, a set of MediaStreams which the transceiver is associated with (allowing LS group assignments), and a set of encodings for the media (used for simulcast as desccribed in sec.simulcast). </ins> 4.1.4.


Question:
Talk about what to do with header extensions in a remote description Expand discussion of how to handle payload types in a remote description Fail things we don’t know about in an answer ,
LGTM
NAME PTAL at my responses
URL If any indicated payload type is unknown, it MUST be ignored. [TODO: should fail on answers] (q: does this logic need to also include processing for other lines - extmap, rtcp-fb, etc - that weren't present in the offer?)
Fixed by
URL 2) * For each specified payload type that is also supported by the local implementation, establish a mapping between the payload type ID and the actual media format. [TODO - Justin to add more to explain mapping.]
Relevant to as well
Fixed by
URL 1) If the media section proto value indicates use of RTP: [TODO: header extensions]
LGTM

For each specified RTP header extension, establish a mapping between the extension ID and URI, as described in section 6 of <del> RFC5285.  If any indicated RTP header extension is unknown, this MUST result in an error. </del> <ins> RFC5285.  If any indicated RTP header extension is not supported, this MUST result in an error. </ins> If the MID header extension is supported, prepare to demux RTP data intended for this media section based on the MID header extension, as described in I-D.ietf-mmusic-msid, Section 3.2. <del> For each specified payload type, establish a mapping between the payload type ID and the actual media format, as described in RFC3264.  If any indicated payload type is unknown, this MUST result in an error. </del> <ins> For each specified media format, establish a mapping between the payload type and the actual media format, as described in RFC3264, Section 6.1.  If any indicated media format is not supported, this MUST result in an error. </ins> For each specified "rtx" media format, establish a mapping between the RTX payload type and its associated primary payload


Question:
Talk about what to do with header extensions in a remote description Expand discussion of how to handle payload types in a remote description Fail things we don’t know about in an answer ,
LGTM
NAME PTAL at my responses
URL If any indicated payload type is unknown, it MUST be ignored. [TODO: should fail on answers] (q: does this logic need to also include processing for other lines - extmap, rtcp-fb, etc - that weren't present in the offer?)
Fixed by
URL 2) * For each specified payload type that is also supported by the local implementation, establish a mapping between the payload type ID and the actual media format. [TODO - Justin to add more to explain mapping.]
Relevant to as well
Fixed by
URL 1) If the media section proto value indicates use of RTP: [TODO: header extensions]
LGTM

If the media section proto value indicates use of RTP: <del> [TODO: header extensions] </del> If the m= section is being recycled (see sec.subsequent- offers), dissociate the currently associated RtpTransceiver by setting its mid attribute to null.


Question:
Talk about what to do with header extensions in a remote description Expand discussion of how to handle payload types in a remote description Fail things we don’t know about in an answer ,
LGTM
NAME PTAL at my responses
URL If any indicated payload type is unknown, it MUST be ignored. [TODO: should fail on answers] (q: does this logic need to also include processing for other lines - extmap, rtcp-fb, etc - that weren't present in the offer?)
Fixed by
URL 2) * For each specified payload type that is also supported by the local implementation, establish a mapping between the payload type ID and the actual media format. [TODO - Justin to add more to explain mapping.]
Relevant to as well
Fixed by
URL 1) If the media section proto value indicates use of RTP: [TODO: header extensions]
LGTM

section by setting the value of the RtpTransceiver's mid attribute to the MID of the m= section. <del> For each specified payload type that is also supported by the local implementation, establish a mapping between the payload type ID and the actual media format.  [TODO - Justin to add more to explain mapping.]  If any indicated payload type is unknown, it MUST be ignored.  [TODO: should fail on answers] </del> <ins> For each specified media format that is also supported by the local implementation, establish a mapping between the specified payload type and the media format, as described in RFC3264, Section 6.1.  Specifically, this means that the implementation records the payload type to be used in outgoing RTP packets when sending each specified media format, as well as the relative preference for each format that is indicated in their ordering.  If any indicated media format is not supported by the local implementation, it MUST be ignored. </ins> For each specified "rtx" media format, establish a mapping between the RTX payload type and its associated primary payload <del> type, as described in RFC4588.  If any referenced primary payload types are not present, this MUST result in an error. </del> <ins> type, as described in RFC4588, Section 4.  If any referenced primary payload types are not present, this MUST result in an error. </ins> For each specified fmtp parameter that is supported by the <del> local implementation, enable them on the associated payload types. </del> <ins> local implementation, enable them on the associated media formats. For each specified RTP header extension that is also supported by the local implementation, establish a mapping between the extension ID and URI, as described in RFC5285, Section 5. Specifically, this means that the implementation records the extension ID to be used in outgoing RTP packets when sending each specified header extension.  If any indicated RTP header extension is not supported by the local implementation, it MUST be ignored. </ins> For each specified RTCP feedback mechanism that is supported by <del> the local implementation, enable them on the associated payload types. </del> <ins> the local implementation, enable them on the associated media formats. </ins> For any specified "TIAS" bandwidth value, set this value as a constraint on the maximum RTP bitrate to be used when sending


Question:
Talk about what to do with header extensions in a remote description Expand discussion of how to handle payload types in a remote description Fail things we don’t know about in an answer ,
LGTM
NAME PTAL at my responses
URL If any indicated payload type is unknown, it MUST be ignored. [TODO: should fail on answers] (q: does this logic need to also include processing for other lines - extmap, rtcp-fb, etc - that weren't present in the offer?)
Fixed by
URL 2) * For each specified payload type that is also supported by the local implementation, establish a mapping between the payload type ID and the actual media format. [TODO - Justin to add more to explain mapping.]
Relevant to as well
Fixed by
URL 1) If the media section proto value indicates use of RTP: [TODO: header extensions]
LGTM

[TODO: handling of CN, telephone-event, "red"] <del> If the media section if of type audio: </del> <ins> If the media section is of type audio: </ins> For any specified "ptime" value, configure the available <del> payload types to use the specified packet size.  If the specified size is not supported for a payload type, use the </del> <ins> media formats to use the specified packet size.  If the specified size is not supported for a media format, use the </ins> next closest value instead. Finally, if this description is of type "pranswer" or "answer",