- From: Justin Uberti <juberti@google.com>
- Date: Thu, 9 Feb 2012 14:38:52 -0500
- To: Jim Barnett <Jim.Barnett@genesyslab.com>
- Cc: rtcweb@ietf.org, public-webrtc@w3.org
- Message-ID: <CAOJ7v-0jMutFbN3b4aC9LQ4BmefwR=tMqp-bYU0vE38cwX548w@mail.gmail.com>
On Thu, Feb 9, 2012 at 10:26 AM, Jim Barnett <Jim.Barnett@genesyslab.com>wrote: > Justin,**** > > What happens if the application JavaScript modifies the offer? Take the > offer generation in example 7.1:**** > > **1. **OffererJS->OffererUA: pc = new PeerConnection();**** > > **2. **OffererJS->OffererUA: pc.addStream(localStream, null);**** > > **3. **OffererJS->OffererUA: pc.startIce();**** > > **4. **OffererUA->OffererJS: iceCallback(candidate, false);**** > > **5. **OffererJS->OffererUA: offer = pc.createOffer(null);**** > > **6. **OffererJS->OffererUA: pc.setLocalDescription(SDP_OFFER, > offer.toSdp());**** > > **7. **OffererJS->AnswererJS: {"type":"OFFER", "sdp":"<offer>"}**** > > ** ** > > Could the JS modify the offer between steps 5 and 6 or between steps 6 and > 7? > Yes for 5/6, but "not recommended" for 6/7. Otherwise, the UA will be out of sync with what was signaled to the remote side. > For example, between steps 5 and 6, could the JS add a stream for which > pc.addStream had not been called previously? Would setLocalDescription > make the necessary modifications in that case? > Since streams need to be coupled to actual media sources, this wouldn't work. (Or if it did, the remote side would just get a stream that never had any media). > What about changes after setLocalDescription has been called? Are > they not allowed? If they are allowed, how does the peer connection know > what to expect from the other side? > You would need to call setLocalDescription again with the modified description in order to ensure the UA is in sync with what the other side will send. > **** > > ** ** > > **- **Jim**** > > ** ** > > ** ** > > ** ** > > *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On > Behalf Of *Justin Uberti > *Sent:* Wednesday, February 08, 2012 3:57 PM > *To:* rtcweb@ietf.org; public-webrtc@w3.org > *Subject:* [rtcweb] New JSEP draft posted**** > > ** ** > > Now available at http://www.ietf.org/id/draft-uberti-rtcweb-jsep-01.txt*** > * > > ** ** > > Includes changes based on implementation feedback, and I believe it > addresses most of the concerns raised in last week's interim meetings:**** > > - Initial documentation provided for each API call, and what state changes > it causes**** > > - More examples, including a complete basic sample application**** > > - Simplified approach to trickle candidate handling**** > > - Resolved concerns about how to separate SDP attributes into media / > transport**** > > - Provided encapsulation for SDP blobs and ICE candidate lines, in the > event we want to change encodings or provide helper functions for JS**** > > - Provided mechanism for limiting candidates to TURN-only**** > > - Resolved several implementation issues**** > > ** ** > > I have not yet addressed the non-overlapping codec concern mentioned in > the interim meeting. I think there are ways of handling this either in the > application or the implementation, but I wanted to get this -01 out first > for feedback.**** >
Received on Thursday, 9 February 2012 19:39:39 UTC