Subject: Re: [MMUSIC] [rtcweb] RTP demux in JSEP and BUNDLE

From: Bernard Aboba <[email-masked]>
Date: Wed, 4 Jan 2017 10:59:03 -0800

I would like to emphasize the importance of standardizing "routing rule"
behavior at an appropriate level of detail.

Developers are encountering interoperability problems and are filing bugs
against implementations.

To address the issues we are seeing, it is necessary to get to the level of
detail in the JSEP-17 Section 6 text.

On Fri, Dec 9, 2016 at 10:01 AM, Bernard Aboba <[email-masked]>
wrote:

> I agree with Eric and Cullen that a more explicit algorithm is needed.
> The proposed text is not specific enough to resolve observed implementation
> differences.
>
> Cullen said:
>
> " I'd also like it be clear that MID takes priority over PT."
>
> [BA] Yes.  This is something that multiple implementations appear to agree
> on.  Also, Peter's suggestion that a MID mismatch result in a packet drop
> should be restored.
>
> "I believe we agreed that if the PT was unique, then it would be used for
> demux as well."
>
> [BA] That is my understanding as well.
>
> However, I'd also like to understand what happens if the PT is not
> unique.  The proposed text does not provide enough detail and existing
> implementations differ in how they treat non-unique PTs. I am familiar with
> an implementation that declares SDP with non-unique PTs to be in error.
> Another implementation fills a Payload Type table with a single value, with
> the value selected depending on whether SSRCs are also specified (e.g.
> specification of SSRCs is taken as an indication that only SSRC matching is
> desired, and therefore that a Payload Type entry is not needed).  The ORTC
> API spec says to latch the SSRC on a Payload Type match and then remove the
> Payload Type table entry, but some implementations have found this leads to
> problems so they have foresaken either the latching or the PT removal (or
> both).
>
> Cullen said:
>
> "I prefer the much more explicit algorithm particularly for the RTCP
> handling as implementors have a hard time figuring out wha they need to do
> to process the RTCP."
>
> [BA] I would also prefer that we have a more explicit algorithm here,
> because we have seen very substantial implementation differences.  One
> implementation I'm familiar with sends the RTCP packets to all RtpSender
> and RtpReceiver objects.  While this is inefficient, it does avoid sending
> RTCP packets to the wrong objects.  Another implementation sorts the RTCP
> reports as indicated in the proposed text - but has found that the required
> handling is so message-dependent that it leads to bugs (e.g. FIR and APP
> messages have caused issues in particular).  So this approach is more
> efficient, unless we are willing to get into detail on every RTCP message,
> it will fall short.
>
>
> On Fri, Dec 9, 2016 at 9:13 AM, Eric Rescorla <[email-masked]> wrote:
>
>> I agree with Cullen that the algorithm structure used in current JSEP is
>> the better structure.
>>
>> Magnus, Colin, do you think you could rewrite your text in that structure?
>>
>> -Ekr
>>
>>
>> On Fri, Dec 9, 2016 at 7:10 AM, Cullen Jennings <[email-masked]> wrote:
>>
>>>
>>> In general, I think it makes sense to put most the details in bundle and
>>> overview in jsep as you have proposed here.
>>>
>>> I believe we agreed that if the PT was unique, then that would be used
>>> for the demux as well. I'd like to see that explicitly spelled out in this
>>> text instead of just left as an option allowed but not really specified by
>>> this text.  I'd also like it be clear that MID takes priority over PT.
>>>
>>> I prefer the much more explicit algorithm particularly for the RTCP
>>> handling as implementors have a hard time figuring out wha they need to do
>>> to process the RTCP.
>>>
>>>
>>>
>>> > On Dec 8, 2016, at 3:54 AM, Colin Perkins <[email-masked]> wrote:
>>> >
>>> > Hi,
>>> >
>>> > [cc’ing RTCWEB and MMUSIC, since this relates to both JSEP and BUNDLE
>>> drafts]
>>> >
>>> > There’s been a lot of discussion on the lists, and at the meeting in
>>> Seoul, around how RTP streams are mapped onto higher-level, application
>>> meaningful, semantic roles. In particular, around how RTP streams map onto
>>> JSEP objects for WebRTC. Magnus Westerlund and I would like to propose the
>>> following updates to JSEP and BUNDLE to try to clarify the behaviour.
>>> >
>>> > Comments and feedback very welcome.
>>> >
>>> > Cheers,
>>> > Colin
>>> >
>>> >
>>> >
>>> > # Replacement for Section 6 of draft-ietf-rtcweb-jsep-17
>>> >
>>> > <section title="Processing RTP/RTCP" anchor="sec.rtp.demux">
>>> >    <t>As described in <xref target="RFC3550"/>, RTP packets are
>>> >    associated with RTP streams <xref target="RFC7656"/>. Metadata
>>> >    about those streams, including source description information
>>> >    and reception quality feedback is conveyed in RTCP packets.
>>> >    Each RTP stream is identified by an SSRC, and each RTP packet
>>> >    carries an SSRC value that is used to associate the packet with
>>> >    the correct RTP stream. RTCP packets also use SSRCs to identify
>>> >    the RTP streams that the reports and metadata relate to.  RTCP
>>> >    packets generally carry multiple SSRC values and report on, or
>>> >    deliver source description information relating to, several RTP
>>> >    streams.</t>
>>> >
>>> >    <t>Each incoming RTP stream, identified by its SSRC, is mapped to
>>> >    an m= section in the SDP. The SDP m= sections then correspond to
>>> >    RtpReceiver objects. This allows each RTP stream to be associated
>>> >    with an RtpTransceiver. Further processing of the RTP stream can
>>> >    then be done at the RtpTransceiver level.  This includes using
>>> >    RID <xref target="I-D.ietf-mmusic-rid"/> to distinguish between
>>> >    multiple Encoded Streams, as well as determine which Source RTP
>>> >    stream should be repaired by a given Redundancy RTP stream.</t>
>>> >
>>> >    <t>The process of mapping RTP streams onto m= sections depends on
>>> >    whether streams are bundled or not. If the SDP BUNDLE extension
>>> >    is in use, then RTP streams are mapped onto m= sections based on
>>> >    the MID values as described in
>>> >    <xref target="I-D.ietf-mmusic-sdp-bundle-negotiation"/>.  If the
>>> >    SDP BUNDLE extension is not in use, each m= section corresponds
>>> >    to a transport layer connection and the RTP streams received on
>>> >    that connection correspond to the m= section.</t>
>>> >
>>> >    <t>Incoming RTCP packets contain metadata including reception
>>> >    quality feedback, source description information, and other
>>> >    signalling relating to RTP streams. The RTCP packets are parsed,
>>> >    the associated RTP streams are identified based on the included
>>> >    SSRC values, and the metadata relating to those RTP streams is
>>> >    updated (this might include updating the MID information, used
>>> >    to associate RTP streams with m= sections, if the SDP BUNDLE
>>> >    extension is in use). This updated metadata is available to the
>>> >    RtpTransceiver objects associated with those RTP streams.
>>> >    </t>
>>> > </section>
>>> >
>>> >
>>> >
>>> > # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-n
>>> egotiation-36
>>> >
>>> >     <section anchor="sec-rtp-pt"
>>> >              title="Associating RTP Streams With Correct SDP Media
>>> Description"
>>> >              toc="default">
>>> >       <t>As described in <xref format="default" pageno="false"
>>> >       target="RFC3550"/>, RTP packets are associated with RTP streams
>>> <xref
>>> >       format="default" pageno="false" target="RFC7656"/>. Each RTP
>>> stream is
>>> >       identified by an SSRC value, and each RTP packet carries an SSRC
>>> value
>>> >       that is used to associate the packet with the correct RTP
>>> stream. RTCP
>>> >       packets also uses SSRCs to identify on which RTP streams any
>>> report or
>>> >       feedback relate to. Thus, an RTCP packet will commonly carry
>>> multiple
>>> >       SSRC values, and might therefore be providing feedback or report
>>> on
>>> >       multiple RTP streams. </t>
>>> >
>>> >       <t>In order to be able to process received RTP packets correctly
>>> it
>>> >       must be possible to associate an RTP stream with the correct "m="
>>> >       line, as the "m=" line and SDP attributes associated with the
>>> "m="
>>> >       line contain information needed to process the packets.</t>
>>> >
>>> >       <t>As all RTP streams associated with a BUNDLE group are part of
>>> the
>>> >       same RTP session and using the same address:port combination for
>>> >       sending and receiving RTP/RTCP packets, the local address:port
>>> >       combination cannot be used to associate an RTP stream with the
>>> correct
>>> >       "m=" line. In addition, multiple RTP streams might be associated
>>> with
>>> >       the same "m=" line.</t>
>>> >
>>> >       <t>Also, as described in <xref format="default" pageno="false"
>>> >       target="sec-rtp-sessions-pt"/>, the same payload type value
>>> might be
>>> >       used by multiple RTP streams, in which case the payload type
>>> value
>>> >       cannot be used to associate an RTP stream with the correct "m="
>>> line.
>>> >       However, there are cases where each "m=" line has unique payload
>>> type
>>> >       values, and then the payload type could serve as hint to the
>>> relevant
>>> >                   "m=" line the RTP stream is associated with.</t>
>>> >
>>> >       <t>An offerer and answerer can inform each other which SSRC
>>> values
>>> >       they will use for an RTP stream by using the SDP 'ssrc' attribute
>>> >       <xref format="default" pageno="false" target="RFC5576"/>.
>>> However, an
>>> >       offerer will not know which SSRC values the answerer will use
>>> until
>>> >       the offerer has received the answer providing that information.
>>> Due to
>>> >       this, before the offerer has received the answer, the offerer
>>> will not
>>> >       be able to associate an RTP stream with the correct "m=" line
>>> using
>>> >       the SSRC value associated with the RTP stream. In addition, the
>>> >       offerer and answerer may start using new SSRC values mid-session,
>>> >       without informing each other using the SDP 'ssrc' attribute.</t>
>>> >
>>> >       <t>In order for an offerer and answerer to always be able to
>>> associate
>>> >       an RTP stream with the correct "m=" line, the offerer and
>>> answerer
>>> >       using the BUNDLE extension MUST support the mechanism defined in
>>> <xref
>>> >       format="default" pageno="false" target="sec-receiver-id"/>,
>>> where the
>>> >       offerer and answerer includes the identification-tag (provided
>>> by the
>>> >       remote peer) associated with an "m=" line in the RTP Streams and
>>> in
>>> >       RTCP SDES packets part of a BUNDLE group.</t>
>>> >
>>> >       <t>The mapping from an SSRC to an identification-tag is carried
>>> in
>>> >       RTCP SDES packets or in RTP header extensions (<xref
>>> format="default"
>>> >       pageno="false" target="sec-receiver-id"/>). Since a compound RTCP
>>> >       packet can contain multiple RTCP SDES packets, and each RTCP SDES
>>> >       packet can contain multiple chunks, an RTCP packet can contain
>>> several
>>> >       SSRC to identification-tag mappings. The offerer and answerer
>>> maintain
>>> >       tables mapping RTP streams identified by SSRC to "m=" lines
>>> identified
>>> >       by the identification-tag.
>>> >       When receiving an RTP packet carrying a MID header extension
>>> >       with the identification-tag, or an RTCP packet carrying one or
>>> >       more SDES MID items, the offerer or answerer creates a mapping
>>> >       table entry between the SSRC value and the identification-tag,
>>> >       in order to associate the RTP stream associated with that SSRC
>>> >       value with the "m=" line corresponding to the
>>> identification-tag.</t>
>>> >
>>> >       <t>The mapping between the SSRC an identification-tag might
>>> change
>>> >       mid-session if, for a given SSRC value, a different
>>> identification-tag
>>> >       is provided in an RTP or RTCP packet. In that case these tables
>>> are
>>> >       updated each time an RTP/RTCP packet containing a new mappings
>>> from
>>> >       SSRC to identification-tag is received. Some considerations for
>>> >       avoiding update flaps are provided in Section 4.2.6 of <xref
>>> >       target="RFC7941"/> which should be followed. </t>
>>> >
>>> >       <t>If an offerer and answerer is not able to associate an RTP
>>> stream
>>> >       with an "m=" line (using the mechanisms described in this
>>> section, or
>>> >       using other appropriate mechanism, e.g., based on the payload
>>> type
>>> >       value if it is unique to a single "m=" line), it MUST either
>>> drop the
>>> >       RTP packets associated with the RTP stream, or process them in an
>>> >       application specific manner, once non-stream specific processing
>>> >       (e.g., related to congestion control) of the RTP packets have
>>> >       occurred.</t>
>>> >
>>> >       <t>When compound RTCP packets are received, they are split
>>> >       into their component RTCP packets and those component RTCP
>>> >       packets are processed based on their RTCP packet type, in
>>> >       the order in which they were placed into the compound RTCP
>>> >       packet. Non-compound RTCP packets are processed based on
>>> >       their RTCP packet type, in the order they are received.
>>> Information
>>> >       in each RTCP packet can relate to one or more RTP streams.
>>> >       For example, RTCP Sender Report (SR) and Receiver Report (RR)
>>> >       packets include an SSRC of sender field that indicates the
>>> >       identity of the participant that sent the RTCP packet, along
>>> >       with a list of Report Blocks. Each report contains data on the
>>> >       reception quality of a single RTP stream, identified by SSRC,
>>> >       as received by the SSRC that sent the RTCP packet. Other RTCP
>>> >       packet types similarly contain references to the SSRC of the
>>> >       sender of the RTCP packet, and the RTP streams to which it
>>> >       refers.</t>
>>> >
>>> >       <t>It should always be possible to process RTCP packets, and
>>> >       store the received information in a data structure associated
>>> >       with an RTP stream, identified by SSRC, for later access and
>>> >       use. It is possible that RTCP packets relating to an SSRC can
>>> >       be received before RTP packets relating to that SSRC, so the
>>> >       data structures relating to an SSRC might need to be created
>>> >       before the corresponding RTP stream is received.</t>
>>> >
>>> >       <t>Similarly, information relating to an RTP stream might be
>>> >       received before the data needed to map it onto an m= line is
>>> >       received. Information carried in RTCP packets relating to such
>>> >       an RTP stream that is application and/or "m=" line dependent
>>> >       MAY be dropped until the SSRCs is associated with a particular
>>> >       "m=" line. However, information to generate RTCP report blocks
>>> >       and other basic transport level feedback or reporting needs to
>>> >       be retained, so RTCP reports relating to the stream can be
>>> >       generated.</t>
>>> >
>>> >     </section>
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Colin Perkins
>>> > https://csperkins.org/
>>> >
>>> >
>>> >
>>> >
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> [email-masked]
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> [email-masked]
>> https://www.ietf.org/mailman/listinfo/mmusic
>>
>>
>

---

From: Eric Rescorla <[email-masked]>
Date: Thu, 5 Jan 2017 10:56:33 -0800

This discussion seems to have stalled. I think it's clear that:

- There's a strong (though not necessarily unanimous) feeling on the JSEP
side that we need to specify an actual algorithm in the way that JSEP did.
- Magnus and Colin feel like the algorithm in JSEP isn't adequate.

Given that, I think the best way to proceed is to schedule a virtual
interim to jointly produce text in algorithmic form that addresses Colin
and Magnus's concerns. This approach worked well the last time we had some
contentious wordsmithing (the rules around IP address handling). Chairs, do
you think you could arrange this sometime soon so it doesn't hold up JSEP.

-Ekr

On Wed, Jan 4, 2017 at 10:59 AM, Bernard Aboba <[email-masked]>
wrote:

> To address the issues we are seeing, it is necessary to get to the level
> of detail in the JSEP-17 Section 6 text.
>
>> [BA] Yes.  This is something that multiple implementations appear to
>> agree on.  Also, Peter's suggestion that a MID mismatch result in a packet
>> drop should be restored.
>>
>>>> In general, I think it makes sense to put most the details in bundle
>>>> and overview in jsep as you have proposed here.
>>>>
>>>> >       identified by an SSRC value, and each RTP packet carries an
>>>> SSRC value
>>>> >       SSRC values, and might therefore be providing feedback or
>>>> report on
>>>> >       <t>In order to be able to process received RTP packets
>>>> correctly it
>>>> >       <t>As all RTP streams associated with a BUNDLE group are part
>>>> of the
>>>> >       "m=" line. In addition, multiple RTP streams might be
>>>> associated with
>>>> >       the same "m=" line.</t>
>>>> >
>>>> >       However, there are cases where each "m=" line has unique
>>>> payload type
>>>> >       using the BUNDLE extension MUST support the mechanism defined
>>>> in <xref
>>>> >       remote peer) associated with an "m=" line in the RTP Streams
>>>> and in

---

From: Peter Thatcher <[email-masked]>
Date: Fri, 06 Jan 2017 19:23:50 +0000

When I read your text, I get the impression that you want to simplify the
algorithm to only supporting MIDs for demux with SSRC latching.  No PT
demux.   No SSRC demux (except for ones latched by MID).   In other words,
if you want to use BUNDLE, you must use MID and only MID.

But it's not clear if that's the case.  Perhaps I missed something.  I
agree with others that a more explicit algorithm would make your intention
clear.

To help, here is my interpretation of your text as an algorithm.  Please
let me know if it's a correct interpretation:

    <section title="Appendix B" anchor="sec.appendix-b">
      <t>To prepare for demultiplexing RTP packets to the correct "m="
      line, the following steps MUST be followed for each BUNDLE
      group.</t>

      <t>
        <list>
          <t>Construct a table mapping MID to "m=" line for each "m="
          line in this BUNDLE group.</t>

          <t>Construct an empty table mapping incoming SSRC to "m=" line.
</t>
        </list>
      </t>

      <t>As "m=" lines are added or removed from the BUNDLE groups, or
      their configurations are changed, the tables above must also be
      updated.</t>

      <t>For each RTP packet received, the following steps MUST be
      followed to route the packet to the correct "m=" section within
      a BUNDLE group:</t>

      <t>
        <list>
          <t>If the packet has a MID and that MID is not in the table
          mapping MID to "m=" line, drop the packet and stop.</t>

          <t>If the packet has a MID and that MID is in the table
          mapping MID to "m=" line, update the incoming SSRC mapping
          table to include an entry that maps the packet's SSRC to the
          "m=" line for that MID.</t>

          <t>If the packet's SSRC is in the incoming SSRC mapping
          table, route the packet to the associated "m=" line and
          stop.</t>

          <t>Otherwise, drop the packet.</t>
        </list>
      </t>

... similar for RTCP ...
    </section>

On Thu, Dec 8, 2016, 2:55 AM Colin Perkins <[email-masked]> wrote:

Hi,

There’s been a lot of discussion on the lists, and at the meeting in Seoul,
around how RTP streams are mapped onto higher-level, application
meaningful, semantic roles. In particular, around how RTP streams map onto
JSEP objects for WebRTC. Magnus Westerlund and I would like to propose the
following updates to JSEP and BUNDLE to try to clarify the behaviour.

Cheers,
Colin

# Replacement text for section 10.2 of
draft-ietf-mmusic-sdp-bundle-negotiation-36

     <section anchor="sec-rtp-pt"
              title="Associating RTP Streams With Correct SDP Media
Description"
              toc="default">
       <t>As described in <xref format="default" pageno="false"
       target="RFC3550"/>, RTP packets are associated with RTP streams <xref
       format="default" pageno="false" target="RFC7656"/>. Each RTP stream
is
       identified by an SSRC value, and each RTP packet carries an SSRC
value
       that is used to associate the packet with the correct RTP stream.
RTCP
       packets also uses SSRCs to identify on which RTP streams any report
or
       feedback relate to. Thus, an RTCP packet will commonly carry multiple
       SSRC values, and might therefore be providing feedback or report on
       multiple RTP streams. </t>

       <t>In order to be able to process received RTP packets correctly it
       must be possible to associate an RTP stream with the correct "m="
       line, as the "m=" line and SDP attributes associated with the "m="
       line contain information needed to process the packets.</t>

       <t>As all RTP streams associated with a BUNDLE group are part of the
       same RTP session and using the same address:port combination for
       sending and receiving RTP/RTCP packets, the local address:port
       combination cannot be used to associate an RTP stream with the
correct
       "m=" line. In addition, multiple RTP streams might be associated with
       the same "m=" line.</t>

       <t>Also, as described in <xref format="default" pageno="false"
       target="sec-rtp-sessions-pt"/>, the same payload type value might be
       used by multiple RTP streams, in which case the payload type value
       cannot be used to associate an RTP stream with the correct "m=" line.
       However, there are cases where each "m=" line has unique payload type
       values, and then the payload type could serve as hint to the relevant
                    "m=" line the RTP stream is associated with.</t>

       <t>An offerer and answerer can inform each other which SSRC values
       they will use for an RTP stream by using the SDP 'ssrc' attribute
       <xref format="default" pageno="false" target="RFC5576"/>. However, an
       offerer will not know which SSRC values the answerer will use until
       the offerer has received the answer providing that information. Due
to
       this, before the offerer has received the answer, the offerer will
not
       be able to associate an RTP stream with the correct "m=" line using
       the SSRC value associated with the RTP stream. In addition, the
       offerer and answerer may start using new SSRC values mid-session,
       without informing each other using the SDP 'ssrc' attribute.</t>

       <t>In order for an offerer and answerer to always be able to
associate
       an RTP stream with the correct "m=" line, the offerer and answerer
       using the BUNDLE extension MUST support the mechanism defined in
<xref
       format="default" pageno="false" target="sec-receiver-id"/>, where the
       offerer and answerer includes the identification-tag (provided by the
       remote peer) associated with an "m=" line in the RTP Streams and in
       RTCP SDES packets part of a BUNDLE group.</t>

       <t>The mapping from an SSRC to an identification-tag is carried in
       RTCP SDES packets or in RTP header extensions (<xref format="default"
       pageno="false" target="sec-receiver-id"/>). Since a compound RTCP
       packet can contain multiple RTCP SDES packets, and each RTCP SDES
       packet can contain multiple chunks, an RTCP packet can contain
several
       SSRC to identification-tag mappings. The offerer and answerer
maintain
       tables mapping RTP streams identified by SSRC to "m=" lines
identified
       by the identification-tag.
       When receiving an RTP packet carrying a MID header extension
       with the identification-tag, or an RTCP packet carrying one or
       more SDES MID items, the offerer or answerer creates a mapping
       table entry between the SSRC value and the identification-tag,
       in order to associate the RTP stream associated with that SSRC
       value with the "m=" line corresponding to the identification-tag.</t>

       <t>The mapping between the SSRC an identification-tag might change
       mid-session if, for a given SSRC value, a different
identification-tag
       is provided in an RTP or RTCP packet. In that case these tables are
       updated each time an RTP/RTCP packet containing a new mappings from
       SSRC to identification-tag is received. Some considerations for
       avoiding update flaps are provided in Section 4.2.6 of <xref
       target="RFC7941"/> which should be followed. </t>

       <t>If an offerer and answerer is not able to associate an RTP stream
       with an "m=" line (using the mechanisms described in this section, or
       using other appropriate mechanism, e.g., based on the payload type
       value if it is unique to a single "m=" line), it MUST either drop the
       RTP packets associated with the RTP stream, or process them in an
       application specific manner, once non-stream specific processing
       (e.g., related to congestion control) of the RTP packets have
       occurred.</t>

       their RTCP packet type, in the order they are received. Information

     </section>

--
Colin Perkins
https://csperkins.org/

---

From: Peter Thatcher <[email-masked]>
Date: Fri, 06 Jan 2017 22:49:14 +0000

I made a PR for JSEP that incorporates your text *and* adds a more specific
algorithm which only uses MID and SSRC latching (my interpretation of your
text, which I'm still not sure is correct):

https://github.com/rtcweb-wg/jsep/pull/489

In practice, I think this is mostly word smithing and doesn't require a
virtual interim.  The big questions, which perhaps to require a virtual
interim, are:

1.  Do we want to include PT demux in the algorithm (when neither SSRC nor
MID match)?
2.  Do we want to include signaled SSRCs in the demux algorithm (the SSRC
table is loaded with signaled SSRCs)?

The question to both of those is "yes" in PR 411 and "no" in PR 489.

---

From: Bernard Aboba <[email-masked]>
Date: Fri, 6 Jan 2017 16:48:02 -0800

Peter said:

"

"

[BA]  I believe that we need to cover both #1 and #2.

1. There have been bugs filed against implementations that relate to PT
demux (e.g. SSRC change scenario).  So IMHO covering PT demux is
necessary.  JSEP-17 was pretty close, but tere are a few issues that still
need to be discussed:
     a.  SSRC latching on a PT match.  If the PT can change without SSRC
changing (e.g. in a video scenario where all codecs have the same
clockrate), it seems that latching could result in mis-routing.
     b. Whether the PT_table is filled if SSRCs are signaled in codecs
and/or RTX/FEC.  Implementations are doing different things here, and that
is going to cause problems.

2. Including signaled SSRCs in the demux algorithm is a practical
requirement because SSRC signaling is widely implemented and MID isn't
yet.  Also, when discussing RTCP segment routing, doesn't the SSRC_table
come into play (e.g. in deciding which receiver to route a Sender Report
to)?

On Fri, Jan 6, 2017 at 2:49 PM, Peter Thatcher <[email-masked]> wrote:

> I made a PR for JSEP that incorporates your text *and* adds a more
> specific algorithm which only uses MID and SSRC latching (my interpretation
> of your text, which I'm still not sure is correct):
>
>> # Replacement text for section 10.2 of draft-ietf-mmusic-sdp-bundle-
>> negotiation-36
>>