Hi Roman, Thank you for your response.
We are using FreeSWITCH as a SIP and RTP media server to connect the caller (leg a) to the callee (leg b), the caller is expecting the media state sendrecv but this is influenced by the 3rd party. The call flow is as follows. -> = generated by FreeSWITCH (caller) <- = generated by 3rd party (callee) -> 0.000000s INVITE with SDP 'codec list' <- 0.012557s 100 Trying <- 0.083254s 200 OK with SDP 'codec list' -> 0.085348s ACK <- 0.071315s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.086461s 100 Trying -> 0.087391s 200 OK with SDP 'codec list' <- 0.154249s ACK with SDP 'codec list and a=sendonly' <- 0.155111s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.166856s 100 Trying -> 0.167631s 200 OK with SDP 'codec list and a=sendonly' <- 0.202331s ACK with SDP 'codec list and a=recvonly' <- 0.337532s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.346448s 100 Trying -> 0.347170s 200 OK with SDP 'codec list and a=sendonly' <- 0.390116s ACK with SDP 'codec list and a=recvonly' FreeSWITCH currently interprets a RE-INVITE with-out SDP for an existing session as 'no change' for the hold state so it's carrying 'a=sendonly' over from the existing session as it was in the ACK SDP generated by the 3rd party. Based on your explanation I believe this is wrong and we should be responding with-out 'a=sendonly' (default behaviour) or with 'a=sendrecv'. Thanks, Shaun ________________________________ From: Roman Shpount <ro...@telurix.com> Sent: 07 September 2020 08:44 To: Shaun Stokes <shaun@sysconfig.cloud> Cc: sip-implementors@lists.cs.columbia.edu <sip-implementors@lists.cs.columbia.edu> Subject: Re: [Sip-implementors] RFC 3261 section 14.2 - "brand new call" does not specify whether the SDP should modify media attributes of an existing session containing a=sendonly or a=recvonly Shaun, I am the person who actually got the language about handing re-INVITE without SDP into RFC 3261 as "a brand new call". The initial intent was to enable a third party call control to initiate a new call by sending a re-INVITE without SDP to an existing call and then place another call to a new party. If I understand correctly, FreeSwitch is sending a response with "a=recvonly" to a re-INVITE with no SDP? If this is the case, since they are a media server, in this particular situation they are probably wrong, but generally the answer is "it depends". Because of this, you are not going to find an RFC that specifies the one and only correct procedure. The general idea is that sendonly/recvonly in every SDP exchange should reflect the preferences for the user agents, not what was previously negotiated. Imagine that one UA is putting another UA on hold. In this case this phone sends a re-INVITE with a=inactive (or a=sendonly which only makes sense if the UA plans to play the music on hold). The second UA will respond with a=inactive or a=recvonly. If the second UA later sends a re-INVITE without SDP, the first UA will still respond with SDP with a=inactive (or a=sendonly), since it is still on hold. If the UA which is currently on hold sends a re-INVITE with no SDP, then the other UA should respond with a=sendrecv (since it is not on hold), but the first UA should respond with a=inactive (or a=sendonly) in SDP in ACK, since it is still on hold. In other words, re-INVITE does not change the local UA hold status, only a user action does this. Based on the local hold status and the remote direction attribute the UA should respond with an appropriate direction attribute in the answer. If you are using FreeSwitch as a media server, then the local call status is likely not on hold and it should be able to send/recv media, which should be indicated in the response to a re-INVITE with no body. In general case, the local call status is something that depends on the application running on FreeSwitch, which you do not specify. This is why the general answer "it depends". I hope it helps, _____________ Roman Shpount On Mon, Sep 7, 2020 at 2:11 AM Shaun Stokes <shaun@sysconfig.cloud> wrote: Hi, We've been struggling with some of the interpretation used in RFC 3261 section 14.2 which has created a conflict between the software we use (FreeSWITCH) and another environment (based on Cisco previously Broadsoft equipment) both of which claim to be RFC 3261 compliant and claim that the other is not. The problem occurs when the 3rd party sends an SDP with the media attribute 'a=sendonly' on an existing session then follow with a RE-INVITE with-out an SDP, they claim that our 2xx offer in response should contain an SDP with-out 'a=sendonly' (or replace with 'a=sendrecv') based on the interpretation of a "brand new call" used below. Anthony Minessale II (FreeSWITCH lead) claims that "brand new call" is only intended to refer to codecs (not all media attributes) and that the 3rd party (Broadsoft) invented this concept on their own. RFC 3261 14.2 UAS Behavior A UAS providing an offer in a 2xx (because the INVITE did not contain an offer) SHOULD construct the offer as if the UAS were making a brand new call, subject to the constraints of sending an offer that updates an existing session, as described in [13] in the case of SDP. Specifically, this means that it SHOULD include as many media formats and media types that the UA is willing to support. The UAS MUST ensure that the session description overlaps with its previous session description in media formats, transports, or other parameters that require support from the peer. This is to avoid the need for the peer to reject the session description. If, however, it is unacceptable to the UAC, the UAC SHOULD generate an answer with a valid session description, and then send a BYE to terminate the session. Source: https://tools.ietf.org/html/rfc3261#section-14.2 The 3rd party have also stated that this isn't a call going on hold as it's routing to an ACD group, according to RFC 6337 section 5.3 "the use of sendonly/recvonly is not limited to hold". In the following discussion on this subject involving the authors of RFC 3261 there is a clear indication that a RE-INVITE with-out an SDP should not modify 'a=sendonly', unfortunately this isn't enough to support our argument and our service providers protocol lead have determined that the 3rd party is acting correctly and have asked for more evidence. http://marc.info/?t=98738614300001&r=1&w=2 We have also pointed out that RFC 3264 section 8 states that the offer MAY be identical to the last SDP provided but are promptly referred back to "brand new call" in RFC 3261 section 14.2. RFC 3264 8 Modifying the Session At any point during the session, either participant MAY issue a new offer to modify characteristics of the session. It is fundamental to the operation of the offer/answer model that the exact same offer/answer procedure defined above is used for modifying parameters of an existing session. The offer MAY be identical to the last SDP provided to the other party (which may have been provided in an offer or an answer), or it MAY be different. We refer to the last SDP provided as the "previous SDP". If the offer is the same, the answer MAY be the same as the previous SDP from the answerer, or it MAY be different. If the offered SDP is different from the previous SDP, some constraints are placed on its construction, discussed below. Source: https://tools.ietf.org/html/rfc3264#section-8 I've been struggling to find anything more in RFC which can support our argument, on the contrary I had a response from Paul Kyzivat on IETF Dispatch who directed me towards RFC 6337. RFC 6337 section 5.1 refers us back to RFC 3261 in case of a RE-INVITE and "without regard for what the other party in the call may have indicated previously" would suggest we should be using 'a=sendrecv' in our offer, Paul's response was a follows. As one of the authors of 6337 I will agree that sendrecv is probably what the UAS should be offering given the circumstances. But it ultimately comes down to what it "wants" to be doing at that time. The folly comes when it offers something less than what *it* wants because it imagines (based on prior o/a) that the answerer wants less than it does. This can get you into "stuck on hold" scenarios or other trouble. This contradicts the previous discission on SIP Implementors ( http://marc.info/?t=98738614300001&r=1&w=2), which behaviour is correct and does the RFC need to be updated? Hope someone here can help. 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