| Summary: | ASTERISK-24242: PJSIP: Error parsing/validating SDP body: Unknown error 220030 | ||
| Reporter: | litnimax (maxgurubot) | Labels: | |
| Date Opened: | 2014-08-17 15:46:57 | Date Closed: | 2014-08-17 16:07:21 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_pjsip |
| Versions: | 12.5.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ( 0) pjsip.conf ( 1) siplog.txt | |
| Description: | When calling from sipjs WebRTP phone the following error message happens. See siplog.txt for details and pjsip.conf for configuration.
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| Comments: | By: litnimax (maxgurubot) 2014-08-17 15:48:06.561-0500 Issue related files. By: Joshua C. Colp (jcolp) 2014-08-17 15:59:14.522-0500 This is not a bug within Asterisk itself. The issue is that PJSIP, by default, builds with a maximum SIP packet size of 4000 bytes. Your message is exceeding this. You can change this by creating a config_site.h file in the pjlib/include/pj directory of your pjproject. In this file place the following: #define PJSIP_MAX_PKT_LEN 8000 You will need to rebuild PJSIP and Asterisk afterwards. By: Matt Jordan (mjordan) 2014-08-17 16:07:11.498-0500 On that same note, that SDP contains multiple video streams. Asterisk can only handle a single media stream of a given type. By: litnimax (maxgurubot) 2014-08-17 16:27:00.715-0500 Thanks much! Did as You advised. But: Asterisk Ready. == Parsing '/home/max/asterisk/a12.5/etc/asterisk/cli.conf': Found *CLI> == WebSocket connection from '127.0.0.1:44671' for protocol 'sip' accepted using version '13' asterisk: ../src/pjsip/sip_transport.c:1607: pjsip_tpmgr_receive_packet: Assertion `rdata->pkt_info.len > 0' failed. Core dumped. Did 'make clean && make' in both pjproject and asterisk folders. By: antonio (antonio) 2015-02-02 16:11:37.069-0600 I am having the same problem, core dump with Assertion `rdata->pkt_info.len > 0' failed. message. How did you fix it? | ||