Reland "Change default secure SCTP protocol to UDP/DTLS/SCTP"

This reverts commit 4436887ed2d3324279e0f2e091c9e9355392721a.

Reason for revert: The original revert was intended to be temporary.

Original change's description:
> Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP"
> 
> This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.
> 
> Reason for revert: Will temporarily revert to fix an issue and reland afterwards.
> 
> Original change's description:
> > Change default secure SCTP protocol to UDP/DTLS/SCTP
> > 
> > The old value - DTLS/SCTP - is not standards conformant,
> > and the new value should be parsable since Chrome M61.
> > 
> > Bug: webrtc:7706
> > Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27940}
> 
> TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:7706
> Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28121}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org,guidou@webrtc.org

Change-Id: I381fa18b644874c20ddaa4cd13fec79a5fd9555a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7706
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139246
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28122}

1 files changed, 1 insertions, 4 deletions

diff --git a/pc/media_session.cc b/pc/media_session.cc
index de96930a0a..1b3d015a79 100644
--- a/pc/media_session.cc
+++ b/pc/media_session.cc
@@ -2236,10 +2236,7 @@ bool MediaSessionDescriptionFactory::AddSctpDataContentForOffer(
   // before we call CreateMediaContentOffer.  Otherwise,
   // CreateMediaContentOffer won't know this is SCTP and will
   // generate SSRCs rather than SIDs.
-  // TODO(deadbeef): Offer kMediaProtocolUdpDtlsSctp (or TcpDtlsSctp), once
-  // it's safe to do so. Older versions of webrtc would reject these
-  // protocols; see https://bugs.chromium.org/p/webrtc/issues/detail?id=7706.
-  data->set_protocol(secure_transport ? kMediaProtocolDtlsSctp
+  data->set_protocol(secure_transport ? kMediaProtocolUdpDtlsSctp
                                       : kMediaProtocolSctp);
   data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp);
   data->set_max_message_size(kSctpSendBufferSize);