| Summary: | ASTERISK-20401: [patch] INFO - RFC2833 transcoding, problem in digits regeneration when there is silence | ||||
| Reporter: | NITESH BANSAL (nbansal) | Labels: | patch | ||
| Date Opened: | 2012-09-10 10:59:50 | Date Closed: | |||
| Priority: | Major | Regression? | No | ||
| Status: | Open/New | Components: | Channels/chan_sip/General Channels/chan_sip/Interoperability | ||
| Versions: | 11.0.0-beta1 13.18.4 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | OS: Debian squeeze distribution x86_64 architecture | Attachments: | ( 0) chan_sip.c_diff.txt ( 1) sip.h_diff.txt ( 2) SIPP_SCRIPTS_AND_SIP_CONF_TO_REPRODUCE_THE_ISSUE.tar.gz | ||
| Description: | "INFO - RFC2833 transcoding". A sends DTMFs using INFO, B Party supports RTPEvent for DTMFs, so asterisk has to regenerate the digits for B side using RTPEvent, The main point of interest in this case that A party sends INFO packets after B has finished playing the audio pcap meaning that A party sends INFO packets when there is silence on the channel between asterisk and B side. Expected Outcome: All digits should be regenerated properly as RTPEvent packets on B side. Actual Outcome: No DTMF End packet is received for last digit | ||||
| Comments: | By: NITESH BANSAL (nbansal) 2012-09-10 11:03:07.492-0500 I have fixed this issue in my code. Please find the diff files for chan_sip.c and sip.h. By: NITESH BANSAL (nbansal) 2012-09-10 11:04:00.344-0500 Please find the sipp scripts and sip.conf to reproduce this issue. | ||||