Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too? First has: audio 10116 RTP/AVP *0* 101 13 Second has: audio 49020 RTP/AVP *8* 13 101 PCAP: http://ge.tt/7MpyBwJ Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..? -Avi
Avi, Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed. A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used. On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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If so.. which I think is not.. is this a bug then in FS? http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP. Voxbeam claims it's not their fault.. is it? -Avi On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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Looking at lines 75-78 you have PCMA enabled on FS as a codec option but not PCMU. When it goes to change to the PCMU codec it looks through the enabled codecs but doesn't find PCMU so hangs up the call with INCOMPATIBLE_DESTINATION. You need to add both PCMA and PCMU in your codec preferences. -Steve On 3 July 2012 14:38, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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Actually rereading the trace, it looks like you have both PCMU & PCMA enabled on lines 39-40. AviMarcus, Can you confirm your codec preferences to be sure? If so, it appears to only be comparing the codec in the 2nd SDP to the codec that's been selected for the call, not to all the codecs enabled on the profile. That does sound like a bug, unless there's some parameter set that's intentionally causing that behaviour... -Steve On 3 July 2012 15:46, Steven Ayre <steveayre@gmail.com> wrote:
Looking at lines 75-78 you have PCMA enabled on FS as a codec option but not PCMU.
When it goes to change to the PCMU codec it looks through the enabled codecs but doesn't find PCMU so hangs up the call with INCOMPATIBLE_DESTINATION.
You need to add both PCMA and PCMU in your codec preferences.
-Steve
On 3 July 2012 14:38, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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Indeed both are enabled... check the PCAP from the original message: http://ge.tt/7MpyBwJ It shows I'm offering in the original invite: audio 28380 RTP/AVP *8 0* 9 101 13 I remember this similar issue with 18x have a different codec than a 200 and Anthony said it was against the RFC but http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp would allow that broken behavior. I'm wondering if this is the same thing if so what RFC to point the vendor to. Would there be a potential negative to turning on this NDLB setting? -Avi On Tue, Jul 3, 2012 at 5:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
Actually rereading the trace, it looks like you have both PCMU & PCMA enabled on lines 39-40. AviMarcus, Can you confirm your codec preferences to be sure?
If so, it appears to only be comparing the codec in the 2nd SDP to the codec that's been selected for the call, not to all the codecs enabled on the profile. That does sound like a bug, unless there's some parameter set that's intentionally causing that behaviour...
-Steve
On 3 July 2012 15:46, Steven Ayre <steveayre@gmail.com> wrote:
Looking at lines 75-78 you have PCMA enabled on FS as a codec option but not PCMU.
When it goes to change to the PCMU codec it looks through the enabled codecs but doesn't find PCMU so hangs up the call with INCOMPATIBLE_DESTINATION.
You need to add both PCMA and PCMU in your codec preferences.
-Steve
On 3 July 2012 14:38, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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Sorry, missed the pcap in the original message. I'd try the NDLB setting and see what happens. Keep in mind what I said about the carrier and FAS... While it's not technically non-compliant behavior it reeks of either incompetence or malice. Generally speaking (in the SDP offer/answer model) as long as the remote end's answer contains codecs in your offer it is valid answer (even if there are several of them). What happens if you remove either PCMA or PCMU from your INVITE (offer)? On Tue, Jul 3, 2012 at 11:02 AM, Avi Marcus <avi@avimarcus.net> wrote:
Indeed both are enabled... check the PCAP from the original message: http://ge.tt/7MpyBwJ It shows I'm offering in the original invite: audio 28380 RTP/AVP 8 0 9 101 13
I remember this similar issue with 18x have a different codec than a 200 and Anthony said it was against the RFC but http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp would allow that broken behavior. I'm wondering if this is the same thing if so what RFC to point the vendor to.
Would there be a potential negative to turning on this NDLB setting?
-Avi
On Tue, Jul 3, 2012 at 5:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
Actually rereading the trace, it looks like you have both PCMU & PCMA enabled on lines 39-40. AviMarcus, Can you confirm your codec preferences to be sure?
If so, it appears to only be comparing the codec in the 2nd SDP to the codec that's been selected for the call, not to all the codecs enabled on the profile. That does sound like a bug, unless there's some parameter set that's intentionally causing that behaviour...
-Steve
On 3 July 2012 15:46, Steven Ayre <steveayre@gmail.com> wrote:
Looking at lines 75-78 you have PCMA enabled on FS as a codec option but not PCMU.
When it goes to change to the PCMU codec it looks through the enabled codecs but doesn't find PCMU so hangs up the call with INCOMPATIBLE_DESTINATION.
You need to add both PCMA and PCMU in your codec preferences.
-Steve
On 3 July 2012 14:38, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle. However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud. What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it? On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you *please* look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite. I think we need Anthony to chime in here... he knows the RFC stuff! -Avi On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
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if its a sip forked dial. We do not support that. On Tue, Jul 3, 2012 at 10:56 AM, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you please look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
18x messages are supposed to make the call unreroutable since they indicate the call has been successful on that route. -Steve On 3 July 2012 16:56, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you *please* look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels
the
call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________
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If we only we had absolutes that were that simple... The default setting in FreeSWITCH when forking (parallel or serial) is to consider the first bridge (leg) that returns media (of any sort) as successful. This FreeSWITCH behavior can be changed. Many other implementations behave differently. In SIP (especially with proxies) this isn't an absolute and several other scenarios are still valid. A leg can return 183 w/SDP and then a 4xx/5xx/6xx, which should then defer to another leg (or pass the error, or branch further, etc). Various scenarios are perfectly valid per the specs. It's up to the implementor/application to decide which behavior is preferred. On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
18x messages are supposed to make the call unreroutable since they indicate the call has been successful on that route.
-Steve
On 3 July 2012 16:56, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you please look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Kristian Kielhofner
So with NDLB-allow-nondup-sdp does that mean when the first SDP chooses PCMA then the second SDP with PCMU comes in - FS will still have that option available? (http://pastebin.freeswitch.org/19423 shows that currently it got lost after the 1st negotiation) This was a parallel dial - the remote opensips apparently was trying various gateways, though.. -Avi On Tue, Jul 3, 2012 at 9:48 PM, Kristian Kielhofner <kris@kriskinc.com>wrote:
If we only we had absolutes that were that simple...
The default setting in FreeSWITCH when forking (parallel or serial) is to consider the first bridge (leg) that returns media (of any sort) as successful. This FreeSWITCH behavior can be changed. Many other implementations behave differently.
In SIP (especially with proxies) this isn't an absolute and several other scenarios are still valid. A leg can return 183 w/SDP and then a 4xx/5xx/6xx, which should then defer to another leg (or pass the error, or branch further, etc). Various scenarios are perfectly valid per the specs. It's up to the implementor/application to decide which behavior is preferred.
On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
18x messages are supposed to make the call unreroutable since they indicate the call has been successful on that route.
-Steve
On 3 July 2012 16:56, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you please look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS
originally
offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <
kris@kriskinc.com>
wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has
been
established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote: > Are multiple 183s from the endpoint that changes the SDP allowed? > I'm > under > the impression this is broken, similar to > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > ... which is why when the codec changes, FS freaks out and cancels > the > call > because of codec negotiation error. > Does that NDLB flag allow this, too? > > First has: audio 10116 RTP/AVP 0 101 13 > Second has: audio 49020 RTP/AVP 8 13 101 > PCAP: http://ge.tt/7MpyBwJ > > Can someone point me to the specific RFC so I can tell the supplier > to > fix > it? > And just curious.. what would make this allowed? A re-INVITE..? > > -Avi > > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Kristian Kielhofner
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Kristian Kielhofner
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True, that is simplistic. I should have said where the 18x isn't ignored... And yes it's one of those situations where everyone's idea of what's right may differ from everyone else's. -Steve On 3 July 2012 19:48, Kristian Kielhofner <kris@kriskinc.com> wrote:
If we only we had absolutes that were that simple...
The default setting in FreeSWITCH when forking (parallel or serial) is to consider the first bridge (leg) that returns media (of any sort) as successful. This FreeSWITCH behavior can be changed. Many other implementations behave differently.
In SIP (especially with proxies) this isn't an absolute and several other scenarios are still valid. A leg can return 183 w/SDP and then a 4xx/5xx/6xx, which should then defer to another leg (or pass the error, or branch further, etc). Various scenarios are perfectly valid per the specs. It's up to the implementor/application to decide which behavior is preferred.
On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre <steveayre@gmail.com> wrote:
18x messages are supposed to make the call unreroutable since they indicate the call has been successful on that route.
-Steve
On 3 July 2012 16:56, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you please look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS
originally
offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <
kris@kriskinc.com>
wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has
been
established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote: > Are multiple 183s from the endpoint that changes the SDP allowed? > I'm > under > the impression this is broken, similar to > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > ... which is why when the codec changes, FS freaks out and cancels > the > call > because of codec negotiation error. > Does that NDLB flag allow this, too? > > First has: audio 10116 RTP/AVP 0 101 13 > Second has: audio 49020 RTP/AVP 8 13 101 > PCAP: http://ge.tt/7MpyBwJ > > Can someone point me to the specific RFC so I can tell the supplier > to > fix > it? > And just curious.. what would make this allowed? A re-INVITE..? > > -Avi > > > >
> Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
-- Kristian Kielhofner
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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-- Kristian Kielhofner
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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I see the original offer from FreeSWITCH. I was asking what the carrier's SBC does if you configure FreeSWITCH to only offer PCMU or PCMA. If you only offer one codec they absolutely shouldn't switch codecs in between 183s... On Tue, Jul 3, 2012 at 11:56 AM, Avi Marcus <avi@avimarcus.net> wrote:
They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you please look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite.
I think we need Anthony to chime in here... he knows the RFC stuff!
-Avi
-- Kristian Kielhofner
Hi guys, I have many potcast mp3 files in a directory: /home/postcast the files beginning with: postcast_year_mon_day_hour_minutes_.mp3 I like to create a playlist in lua: 1. read all files, and make a messagecount. 2. generate a menu to use from phone: press 1 for the next postcast. 2, To repeate the postcast 3. to go to the prev potcast 4 for the fist potcast 5 for the last potcast 6 to hear the date an time for these. I like to sort, firt in first out. I don.t find a command in lua: to read the filelist, to generate a filecount. and to generate a navigation keys: forward, backward, repeate and say the time and date of the current file. Can your help please? nice thanks to all. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc
You'll have a challenge here because natively Lua has no directory commands. I have never tried reading in the results of a system call into a Lua table, but it might be possible. I'd ask on #Lua or check Lua docs to see if such a thing is even possible. Once you have the files in a table then you could do the next/prev/first/last/repeat thing. -MC On Tue, Jul 3, 2012 at 8:58 AM, Thomas Hoellriegel <admin@blindi.net> wrote:
Hi guys, I have many potcast mp3 files in a directory: /home/postcast the files beginning with: postcast_year_mon_day_hour_**minutes_.mp3 I like to create a playlist in lua: 1. read all files, and make a messagecount. 2. generate a menu to use from phone:
press 1 for the next postcast. 2, To repeate the postcast 3. to go to the prev potcast 4 for the fist potcast 5 for the last potcast 6 to hear the date an time for these.
I like to sort, firt in first out.
I don.t find a command in lua: to read the filelist, to generate a filecount. and to generate a navigation keys: forward, backward, repeate and say the time and date of the current file.
Can your help please?
nice thanks to all.
--------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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you can use LuaFileSystem http://keplerproject.github.com/luafilesystem/ On 7/3/2012 5:48 PM, Michael Collins wrote:
You'll have a challenge here because natively Lua has no directory commands. I have never tried reading in the results of a system call into a Lua table, but it might be possible. I'd ask on #Lua or check Lua docs to see if such a thing is even possible. Once you have the files in a table then you could do the next/prev/first/last/repeat thing.
-MC
On Tue, Jul 3, 2012 at 8:58 AM, Thomas Hoellriegel <admin@blindi.net <mailto:admin@blindi.net>> wrote:
Hi guys, I have many potcast mp3 files in a directory: /home/postcast the files beginning with: postcast_year_mon_day_hour_minutes_.mp3 I like to create a playlist in lua: 1. read all files, and make a messagecount. 2. generate a menu to use from phone:
press 1 for the next postcast. 2, To repeate the postcast 3. to go to the prev potcast 4 for the fist potcast 5 for the last potcast 6 to hear the date an time for these.
I like to sort, firt in first out.
I don.t find a command in lua: to read the filelist, to generate a filecount. and to generate a navigation keys: forward, backward, repeate and say the time and date of the current file.
Can your help please?
nice thanks to all.
--------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
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This is another case of what works vs what the rules say and how they should be interpreted. It for sure is not a bug because there are ways to prove its optional. I could be wrong, and like kris an operating from the top of my head but I am pretty sure: If you have 100rel on (which we don't by default because of issues in the sofia stack) then once you answer with a sdp, you must not send anything other than that exact sdp on subsequent responses. The RFC mandates you ignore any sdp that differes from the last. if you DONT have 100rel enabled, they you are allowed to completely ignore the 1xx + sdp messages. So either way we could claim not to support changing the codec in the fashion described but on top of any rules we simply do not do it by default because there are a ton of broken devices who unintentionally send a series of packets that cause us to break. NDLB-allow-nondup-sdp will unlock this to the best we support it. However once we have chosen a codec we will only support that one codec in this mode iirc. You need a new offer and renegotiate-codec-on-reinvite to change the codecs. On Tue, Jul 3, 2012 at 10:22 AM, Kristian Kielhofner <kris@kriskinc.com> wrote:
That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle.
However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud.
What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it?
On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus <avi@avimarcus.net> wrote:
If so.. which I think is not.. is this a bug then in FS?
http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP.
Voxbeam claims it's not their fault.. is it?
-Avi
On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
Avi,
Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed.
A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used.
On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus <avi@avimarcus.net> wrote:
Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too?
First has: audio 10116 RTP/AVP 0 101 13 Second has: audio 49020 RTP/AVP 8 13 101 PCAP: http://ge.tt/7MpyBwJ
Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..?
-Avi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (7)
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Abaci
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Anthony Minessale
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Avi Marcus
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Kristian Kielhofner
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Michael Collins
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Steven Ayre
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Thomas Hoellriegel