Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Daniel Pocock wrote: On 01/04/13 22:06, Joshua Colp wrote: Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: From a code perspective, it could. Still not something I would be comfortable with putting in Asterisk 1.8. http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? Excellent question... I've seen one of my Polycom devices reboot itself each time it receives a raw SDP from WebRTC, so if such a hack is implemented, I'd guess that stripping the F is better than ignoring it. Asterisk doesn't forward SDP through, each leg is completely independent. Without configurability of avpf then if you call a device you have to either not offer it, offer it only, or offer both. Doing both will probably make many devices unhappy, as you have mentioned. So to have stuff really be functional you have to backport AVPF. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 31/03/13 23:43, Joshua Colp wrote: Daniel Pocock wrote: I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new feature it was only added to Asterisk 11. You could try to backport the changes but chan_sip has changed quite a bit, so it could be rough. Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: From a code perspective, it could. Still not something I would be comfortable with putting in Asterisk 1.8. http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 01/04/13 22:06, Joshua Colp wrote: Daniel Pocock wrote: Thanks for the fast reply. I agree backporting full support for AVPF would not be justified for many use cases (including my own). What I was more curious about is whether the F can be tolerated (in other words, ignored or silently removed), as described here: From a code perspective, it could. Still not something I would be comfortable with putting in Asterisk 1.8. http://www.ietf.org/mail-archive/web/rtcweb/current/msg01145.html 1) RTCWEB end-point will always signal AVPF or SAVPF. I signalling gateway to legacy will change that by removing the F to AVP or SAVP. and whether such behavior is possible even without setting avpf=yes on a per-peer basis? This is fine for incoming but what about outgoing to a device? Excellent question... I've seen one of my Polycom devices reboot itself each time it receives a raw SDP from WebRTC, so if such a hack is implemented, I'd guess that stripping the F is better than ignoring it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
On 17/12/12 13:34, Joshua Colp wrote: Barco You wrote: Dear All, Hola, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Insufficient information in SDP (c=)... As the warning states - you haven't enabled AVPF support. This is generally done on a per-peer basis using avpf=yes in the configuration. I would suggest you follow https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support since there may be other things you have missed. I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Also, I'm using a SIP proxy and it takes care of handling all the WebRTC connections and proxying the requests into a normal TCP/TLS connection to Asterisk. I was hoping to avoid opening up WebRTC access directly on Asterisk. One effect this has is that I can't control the `avpf=yes' setting on a per-peer basis, as the proxy is carrying requests from various types of peer, some public, some private. Is there any outright reason Asterisk can't support (S)AVPF on demand? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Daniel Pocock wrote: I'm trying to call from DruCall to Asterisk and I get this error: WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F 103 104 111 0 8 107 106 105 13 126' == Problem setting up ssl connection: error::lib(0):func(0):reason(0) I'm guessing my Asterisk is too old (it is 1.8 from Debian). Can you confirm which version is needed to parse a media descriptor with SAVPF? Do I need to upgrade all the way to v11 with WebRTC support, or was avpf support added in some intermediate version? Asterisk 1.8 does not have any knowledge of AVPF, and since it's a new feature it was only added to Asterisk 11. You could try to backport the changes but chan_sip has changed quite a bit, so it could be rough. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Barco You wrote: Dear All, Hola, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-] chan_sip.c: Insufficient information in SDP (c=)... As the warning states - you haven't enabled AVPF support. This is generally done on a per-peer basis using avpf=yes in the configuration. I would suggest you follow https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support since there may be other things you have missed. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
