[Asterisk-code-review] res pjsip sdp rtp: Properly set packetization on the codec (asterisk[13])

[Sean Bright]

Sean Bright has uploaded a new patch set (#2) to the change originally created by Robert Cripps. ( https://gerrit.asterisk.org/10495 )

Change subject: res_pjsip_sdp_rtp: Properly set packetization on the codec
......................................................................

res_pjsip_sdp_rtp: Properly set packetization on the codec

In the transcode case (Alaw to Ulaw) the ptime was being discarded.

res_rtp_asterisk:

In the native bridge case the framing was never checked.

ASTERISK-28110 #close

Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607
---
M res/res_pjsip_sdp_rtp.c
M res/res_rtp_asterisk.c
2 files changed, 12 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/95/10495/2
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Ic64b8fc6a140a93fbdb2f97550a40d0ff334e607
Gerrit-Change-Number: 10495
Gerrit-PatchSet: 2
Gerrit-Owner: Robert Cripps <rcripps at voxbone.com>
Gerrit-Reviewer: Jenkins2 (1000185)
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
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