Review Board 1.7.16


ARI/PJSIP: Apply requesting channel's capabilities to originated channels in ARI; clean up a bit of PJSIP's usage of format capabalities

Review Request #4434 - Created Feb. 20, 2015 and submitted

Matt Jordan
13
ASTERISK-24812
Reviewers
asterisk-dev
Asterisk
This patch addresses the following problems:
* ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks:
  (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities.
  (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding.
  Fixing this simply entails removing the providing of the format capabilities structure to the Dial API.

* chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio.

* channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following:

PJSIP/A => L;1 <=> L;2 => PJSIP/B
g,u,a     g,u,a    g,u,a      u

That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with:

PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
  g          g   X   u        u

Which causes g722 to be written to PJSIP/B.

Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like:

PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B
  u          u       u         u

We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart.

There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, I punted on improving this and simply updated the comment.

* res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. The comment about bridge_softmix is a purely theoretical concern that should not limit what the PJSIP stack does.
The PJSIP SDP Offer/Answer tests all pass. Prior to this patch, we could set up to 8 transcoding paths on a channel chain created by ARI; now, we have none (if both far ends support the same formats).
Review request changed
Updated (Feb. 24, 2015, 4:24 p.m.)
  • changed from pending to submitted
Committed in revision 432218

https://reviewboard.asterisk.org/ runs on a server provided by Digium, Inc. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA.
Please report problems with this site to asteriskteam@digium.com.