Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote:> Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > > > On 20 April 2017 at 12:42, Jonas Kellens <jonas.kellens at telenet.be> wrote: >> Hello >> >> in sip.conf I have ; >> >> videosupport=yes >> >> >> >> >> Kind regards. >> >> J. >> >> >> >> On 20-04-17 13:09, Marcelo Terres wrote: >>> I suppose that you enable the video support on sip.conf, right? >>> >>> Regards, >>> Marcelo H. Terres <mhterres at gmail.com> >>> IM: mhterres at jabber.mundoopensource.com.br >>> https://www.mundoopensource.com.br >>> https://twitter.com/mhterres >>> https://linkedin.com/in/marceloterres >>> >>> >>> On 19 April 2017 at 13:18, Jonas Kellens <jonas.kellens at telenet.be> wrote: >>>> Hello >>>> >>>> using asterisk 1.8.32.3 >>>> >>>> I am not able to make a call with video support. I do not know what I am >>>> missing to make this video call. >>>> >>>> Codec h264 should be supported. >>>> >>>> >>>> sip*CLI> core show codecs >>>> Disclaimer: this command is for informational purposes only. >>>> It does not indicate anything about your configuration. >>>> INT BINARY HEX TYPE NAME >>>> DESCRIPTION >>>> >>>> ----------------------------------------------------------------------------------- >>>> 1 (1 << 0) (0x1) audio g723 >>>> (G.723.1) >>>> 2 (1 << 1) (0x2) audio gsm >>>> (GSM) >>>> 4 (1 << 2) (0x4) audio ulaw >>>> (G.711 u-law) >>>> 8 (1 << 3) (0x8) audio alaw >>>> (G.711 A-law) >>>> 16 (1 << 4) (0x10) audio g726aal2 >>>> (G.726 AAL2) >>>> 32 (1 << 5) (0x20) audio adpcm >>>> (ADPCM) >>>> 64 (1 << 6) (0x40) audio slin >>>> (16 >>>> bit Signed Linear PCM) >>>> 128 (1 << 7) (0x80) audio lpc10 >>>> (LPC10) >>>> 256 (1 << 8) (0x100) audio g729 >>>> (G.729A) >>>> 512 (1 << 9) (0x200) audio speex >>>> (SpeeX) >>>> 1024 (1 << 10) (0x400) audio ilbc >>>> (iLBC) >>>> 2048 (1 << 11) (0x800) audio g726 >>>> (G.726 RFC3551) >>>> 4096 (1 << 12) (0x1000) audio g722 >>>> (G722) >>>> 8192 (1 << 13) (0x2000) audio siren7 >>>> (ITU >>>> G.722.1 (Siren7, licensed from Polycom)) >>>> 16384 (1 << 14) (0x4000) audio siren14 >>>> (ITU >>>> G.722.1 Annex C, (Siren14, licensed from Polycom)) >>>> 32768 (1 << 15) (0x8000) audio slin16 >>>> (16 >>>> bit Signed Linear PCM (16kHz)) >>>> 65536 (1 << 16) (0x10000) image jpeg >>>> (JPEG >>>> image) >>>> 131072 (1 << 17) (0x20000) image png >>>> (PNG >>>> image) >>>> 262144 (1 << 18) (0x40000) video h261 >>>> (H.261 Video) >>>> 524288 (1 << 19) (0x80000) video h263 >>>> (H.263 Video) >>>> 1048576 (1 << 20) (0x100000) video h263p >>>> (H.263+ Video) >>>> 2097152 (1 << 21) (0x200000) video h264 >>>> (H.264 Video) >>>> 4194304 (1 << 22) (0x400000) video mpeg4 >>>> (MPEG4 Video) >>>> 8388608 (1 << 23) (0x800000) video unknown >>>> (unknown) >>>> 16777216 (1 << 24) (0x1000000) video unknown >>>> (unknown) >>>> 33554432 (1 << 25) (0x2000000) text unknown >>>> (unknown) >>>> 67108864 (1 << 26) (0x4000000) text red >>>> (T.140 Realtime Text with redundancy) >>>> 134217728 (1 << 27) (0x8000000) text t140 >>>> (Passthrough T.140 Realtime Text) >>>> 268435456 (1 << 28) (0x10000000) text unknown >>>> (unknown) >>>> 536870912 (1 << 29) (0x20000000) text unknown >>>> (unknown) >>>> 1073741824 (1 << 30) (0x40000000) (unk) unknown >>>> (unknown) >>>> 2147483648 (1 << 31) (0x80000000) (unk) unknown >>>> (unknown) >>>> 4294967296 (1 << 32) (0x100000000) audio g719 >>>> (ITU >>>> G.719) >>>> 8589934592 (1 << 33) (0x200000000) audio speex16 >>>> (SpeeX 16khz) >>>> 17179869184 (1 << 34) (0x400000000) audio unknown >>>> (unknown) >>>> 34359738368 (1 << 35) (0x800000000) audio unknown >>>> (unknown) >>>> 68719476736 (1 << 36) (0x1000000000) audio unknown >>>> (unknown) >>>> 137438953472 (1 << 37) (0x2000000000) audio unknown >>>> (unknown) >>>> 274877906944 (1 << 38) (0x4000000000) audio unknown >>>> (unknown) >>>> 549755813888 (1 << 39) (0x8000000000) audio unknown >>>> (unknown) >>>> 1099511627776 (1 << 40) (0x10000000000) audio unknown >>>> (unknown) >>>> 2199023255552 (1 << 41) (0x20000000000) audio unknown >>>> (unknown) >>>> 4398046511104 (1 << 42) (0x40000000000) audio unknown >>>> (unknown) >>>> 8796093022208 (1 << 43) (0x80000000000) audio unknown >>>> (unknown) >>>> 17592186044416 (1 << 44) (0x100000000000) audio unknown >>>> (unknown) >>>> 35184372088832 (1 << 45) (0x200000000000) audio unknown >>>> (unknown) >>>> 70368744177664 (1 << 46) (0x400000000000) audio unknown >>>> (unknown) >>>> 140737488355328 (1 << 47) (0x800000000000) audio testlaw >>>> (G.711 test-law) >>>> 281474976710656 (1 << 48) (0x1000000000000) video unknown >>>> (unknown) >>>> 562949953421312 (1 << 49) (0x2000000000000) video unknown >>>> (unknown) >>>> 1125899906842624 (1 << 50) (0x4000000000000) video unknown >>>> (unknown) >>>> 2251799813685248 (1 << 51) (0x8000000000000) video unknown >>>> (unknown) >>>> 4503599627370496 (1 << 52) (0x10000000000000) video unknown >>>> (unknown) >>>> 9007199254740992 (1 << 53) (0x20000000000000) video unknown >>>> (unknown) >>>> 18014398509481984 (1 << 54) (0x40000000000000) video unknown >>>> (unknown) >>>> 36028797018963968 (1 << 55) (0x80000000000000) video unknown >>>> (unknown) >>>> 72057594037927936 (1 << 56) (0x100000000000000) video unknown >>>> (unknown) >>>> 144115188075855872 (1 << 57) (0x200000000000000) video unknown >>>> (unknown) >>>> 288230376151711744 (1 << 58) (0x400000000000000) video unknown >>>> (unknown) >>>> 576460752303423488 (1 << 59) (0x800000000000000) video unknown >>>> (unknown) >>>> 1152921504606846976 (1 << 60) (0x1000000000000000) video unknown >>>> (unknown) >>>> 2305843009213693952 (1 << 61) (0x2000000000000000) video unknown >>>> (unknown) >>>> 4611686018427387904 (1 << 62) (0x4000000000000000) video unknown >>>> (unknown) >>>> >>>> >>>> My 2 video SIP peers : >>>> >>>> >>>> sip*CLI> sip show peer 660081914 >>>> >>>> >>>> * Name : 660081914 >>>> Realtime peer: Yes, cached >>>> Secret : <Set> >>>> MD5Secret : <Not set> >>>> Remote Secret: <Not set> >>>> Context : from-660081 >>>> Subscr.Cont. : <Not set> >>>> Language : nl >>>> Accountcode : 660081intern >>>> AMA flags : Unknown >>>> Transfer mode: open >>>> CallingPres : Presentation Allowed, Not Screened >>>> Callgroup : >>>> Pickupgroup : >>>> MOH Suggest : default >>>> Mailbox : >>>> VM Extension : asterisk >>>> LastMsgsSent : 32767/65535 >>>> Call limit : 2147483647 >>>> Max forwards : 0 >>>> Dynamic : Yes >>>> Callerid : "" <> >>>> MaxCallBR : 384 kbps >>>> Expire : 124 >>>> Insecure : no >>>> Force rport : Yes >>>> ACL : No >>>> DirectMedACL : No >>>> T.38 support : No >>>> T.38 EC mode : Unknown >>>> T.38 MaxDtgrm: 4294967295 >>>> DirectMedia : No >>>> PromiscRedir : No >>>> User=Phone : No >>>> Video Support: Yes >>>> Text Support : No >>>> Ign SDP ver : No >>>> Trust RPID : No >>>> Send RPID : Yes >>>> TrustIDOutbnd: Legacy >>>> Subscriptions: Yes >>>> Overlap dial : No >>>> DTMFmode : rfc2833 >>>> Timer T1 : 500 >>>> Timer B : 32000 >>>> ToHost : >>>> Addr->IP : 1.1.1.1:62850 >>>> Defaddr->IP : (null) >>>> Prim.Transp. : UDP >>>> Allowed.Trsp : UDP >>>> Def. Username: 660081914 >>>> SIP Options : (none) >>>> Codecs : 0x20010a (gsm|alaw|g729|h264) >>>> Codec Order : (alaw:20,g729:20,gsm:20) >>>> Auto-Framing : No >>>> Status : OK (16 ms) >>>> Useragent : MicroSIP/3.15.1 >>>> Reg. Contact : sip:660081914 at 192.168.1.104:62850;ob >>>> Qualify Freq : 120000 ms >>>> Sess-Timers : Refuse >>>> Sess-Refresh : uas >>>> Sess-Expires : 1800 secs >>>> Min-Sess : 90 secs >>>> RTP Engine : asterisk >>>> Parkinglot : >>>> Use Reason : No >>>> Encryption : No >>>> >>>> >>>> >>>> sip*CLI> sip show peer 660081915 >>>> >>>> >>>> * Name : 660081915 >>>> Realtime peer: Yes, cached >>>> Secret : <Set> >>>> MD5Secret : <Not set> >>>> Remote Secret: <Not set> >>>> Context : from-660081 >>>> Subscr.Cont. : <Not set> >>>> Language : nl >>>> Accountcode : 660081intern >>>> AMA flags : Unknown >>>> Transfer mode: open >>>> CallingPres : Presentation Allowed, Not Screened >>>> Callgroup : >>>> Pickupgroup : >>>> MOH Suggest : default >>>> Mailbox : >>>> VM Extension : asterisk >>>> LastMsgsSent : 32767/65535 >>>> Call limit : 2147483647 >>>> Max forwards : 0 >>>> Dynamic : Yes >>>> Callerid : "" <> >>>> MaxCallBR : 384 kbps >>>> Expire : 124 >>>> Insecure : no >>>> Force rport : Yes >>>> ACL : No >>>> DirectMedACL : No >>>> T.38 support : No >>>> T.38 EC mode : Unknown >>>> T.38 MaxDtgrm: 4294967295 >>>> DirectMedia : No >>>> PromiscRedir : No >>>> User=Phone : No >>>> Video Support: Yes >>>> Text Support : No >>>> Ign SDP ver : No >>>> Trust RPID : No >>>> Send RPID : Yes >>>> TrustIDOutbnd: Legacy >>>> Subscriptions: Yes >>>> Overlap dial : No >>>> DTMFmode : rfc2833 >>>> Timer T1 : 500 >>>> Timer B : 32000 >>>> ToHost : >>>> Addr->IP : 1.1.1.1:62812 >>>> Defaddr->IP : (null) >>>> Prim.Transp. : UDP >>>> Allowed.Trsp : UDP >>>> Def. Username: 660081915 >>>> SIP Options : (none) >>>> Codecs : 0x20010a (gsm|alaw|g729|h264) >>>> Codec Order : (alaw:20,g729:20,gsm:20) >>>> Auto-Framing : No >>>> Status : OK (16 ms) >>>> Useragent : MicroSIP/3.15.1 >>>> Reg. Contact : sip:660081915 at 192.168.1.105:62812;ob >>>> Qualify Freq : 120000 ms >>>> Sess-Timers : Refuse >>>> Sess-Refresh : uas >>>> Sess-Expires : 1800 secs >>>> Min-Sess : 90 secs >>>> RTP Engine : asterisk >>>> Parkinglot : >>>> Use Reason : No >>>> Encryption : No >>>> >>>> >>>> >>>> >>>> Kind regards >>>> >>>> >>>> Jonas >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> Check out the new Asterisk community forum at: >>>> https://community.asterisk.org/ >>>> >>>> New to Asterisk? Start here: >>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170421/4d914f5b/attachment-0001.html>
Derek Bolichowski
2017-Apr-21 14:33 UTC
[asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, April 21, 2017 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Subject: Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello
you mean while placing a video call ? What info am I looking for in the debug
output ?
Kind regards.
J.
<snip>
Why not try removing all codecs from the SIP Peer (deny all,
allow only H264), unregister the peer, and try a video call again? If it works,
try adding G711 back but keep H264 at the top of the priority.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20170421/992f3047/attachment.html>
Hello this is the debug output of a test video call. You see codec negotiation but at the end only alaw is chosen and gone is the video ! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our capabilities are 0x20010a (gsm|alaw|g729|h264) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This channel can handle video! HOLLYWOOD next! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: This call needs video offers! [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our capability: 0x20000a (gsm|alaw|h264) Video flag: False Text flag: False [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: -- Done with adding codecs to SDP [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Done building SDP. Settling with this capability: 0x20000a (gsm|alaw|h264) [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Initializing initreq for method INVITE - callid 6ca7f3fb70b56f6f6f9373b776cd495d at 11.22.33.44:5060 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 0 [ 54]: INVITE sip:sipaccount12 at 192.168.1.111:50104;ob SIP/2.0 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 11.22.33.44:5060;branch=z9hG4bK54e24150;rport [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 3 [ 62]: From: "My Account" <sip:71 at 11.22.33.44>;tag=as130bc3f0 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 4 [ 45]: To: <sip:sipaccount12 at 192.168.1.111:50104;ob> [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 5 [ 37]: Contact: <sip:71 at 11.22.33.44:5060> [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 6 [ 61]: Call-ID: 6ca7f3fb70b56f6f6f9373b776cd495d at 11.22.33.44:5060 [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 8 [ 21]: User-Agent: mydomain [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 9 [ 35]: Date: Mon, 26 Jun 2017 12:20:55 GMT [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 10 [ 90]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 11 [ 19]: Supported: replaces [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 12 [ 42]: Alert-Info: <http://127.0.0.1>;info=intern [Jun 26 14:20:55] DEBUG[28609] chan_sip.c: Header 14 [ 29]: Content-Type: application/sdp ... [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 1 [ 94]: Via: SIP/2.0/UDP 11.22.33.44:5060;rport=5060;received=11.22.33.44;branch=z9hG4bK54e24150 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 2 [ 61]: Call-ID: 6ca7f3fb70b56f6f6f9373b776cd495d at 11.22.33.44:5060 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 3 [ 62]: From: "My Account" <sip:71 at 11.22.33.44>;tag=as130bc3f0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 4 [ 76]: To: <sip:sipaccount12 at 192.168.1.111;ob>;tag=8a1f91570e9f434c9da9aca27ded7fb9 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 6 [ 96]: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 7 [ 50]: Contact: <sip:sipaccount12 at 192.168.1.111:50104;ob> [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 8 [ 46]: Supported: replaces, 100rel, timer, norefersub [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 10 [ 19]: Content-Length: 469 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Header 11 [ 0]: [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 0 [ 3]: v=0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 1 [ 46]: o=- 3707475663 3707475664 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 2 [ 9]: s=pjmedia [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 3 [ 8]: b=AS:352 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 4 [ 5]: t=0 0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 5 [ 9]: a=X-nat:0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 6 [ 26]: m=audio 4000 RTP/AVP 8 101 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 7 [ 22]: c=IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 8 [ 12]: b=TIAS:64000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 9 [ 32]: a=rtcp:4001 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 11 [ 20]: a=rtpmap:8 PCMA/8000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 12 [ 33]: a=rtpmap:101 telephone-event/8000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 13 [ 15]: a=fmtp:101 0-16 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 14 [ 23]: m=video 4002 RTP/AVP 99 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 15 [ 22]: c=IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 16 [ 13]: b=TIAS:256000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 17 [ 32]: a=rtcp:4003 IN IP4 192.168.1.111 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 18 [ 10]: a=sendrecv [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 19 [ 22]: a=rtpmap:99 H264/90000 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Body 20 [ 55]: a=fmtp:99 profile-level-id=42000a; packetization-mode=0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: SIP response 200 to standard invite [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP o=- 3707475663 3707475664 IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP b=AS:352... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 8 based on m type on 0x7efde80a5930 [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 101 based on m type on 0x7efde80a5930 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP c=IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP b=TIAS:64000... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4001 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] rtp_engine.c: Setting payload 99 based on m type on 0x7efde80a47f0 [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP c=IN IP4 192.168.1.111... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP b=TIAS:256000... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=rtcp:4003 IN IP4 192.168.1.111... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/90000... OK. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: Processing media-level (video) SDP a=fmtp:99 profile-level-id=42000a; packetization-mode=0... UNSUPPORTED OR FAILED. [Jun 26 14:20:57] DEBUG[1932] chan_sip.c: We're settling with these formats: 0x8 (alaw) Op 21-04-17 om 16:33 schreef Derek Bolichowski:> > *From:*asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Jonas > Kellens > *Sent:* Friday, April 21, 2017 10:18 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > *Subject:* Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264) > > Hello > > > you mean while placing a video call ? What info am I looking for in > the debug output ? > > > > > Kind regards. > > J. > > > <snip> > > Why not try removing all codecs from the SIP Peer (deny all, allow > only H264), unregister the peer, and try a video call again? If it > works, try adding G711 back but keep H264 at the top of the priority. > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170626/7c1a6daf/attachment.html>