Port changes in multiple 183s cause no audio after 200
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff
This sounds like it belongs on Jira so the issue can be tracked. On 27 January 2015 at 00:04, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Hello,
The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp://www.freeswitchsolutions.co m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services: consulting@... http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch.
Michael, I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>. - Jeff On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges < m.stgeorges@csl-technologies.com> wrote:
Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp://www.freeswitchsolutions.co m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services: consulting@... http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Just an observation, but what you wrote in the Jira appears to be different from your original email. 'In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.' So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now? On 16 March 2015 at 14:18, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Michael,
I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.
- Jeff
On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges < m.stgeorges@csl-technologies.com> wrote:
Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp://www.freeswitchsolutions.co m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services: consulting@... http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify? On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre@gmail.com> wrote:
Just an observation, but what you wrote in the Jira appears to be different from your original email.
'In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.'
So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?
On 16 March 2015 at 14:18, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Michael,
I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.
- Jeff
On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges < m.stgeorges@csl-technologies.com> wrote:
Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp://www.freeswitchsolutions.co m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services: consulting@... http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
They're two separate issues. It looks like I referenced the wrong Jira in my reply to Michael's comment. The correct Jira for this issue, involving the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>. Oops. I'll retest shortly. Agreed, the 180-after-183 no audio issue described in FS-7207 <https://freeswitch.org/jira/browse/FS-7207> is clearly different. Sorry for the confusion. - Jeff On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian@freeswitch.org> wrote:
If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify?
On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre@gmail.com> wrote:
Just an observation, but what you wrote in the Jira appears to be different from your original email.
'In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.'
So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?
On 16 March 2015 at 14:18, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Michael,
I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.
- Jeff
On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges < m.stgeorges@csl-technologies.com> wrote:
Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp:// www.freeswitchsolutions.co m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp:// lists.freeswitch.org/mailman/listinfo/freeswi tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services: consulting@... http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org
Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Confirmed, this issue is still present in FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13 21:50:13Z 64bit). - Jeff On Mon, Mar 16, 2015 at 11:58 AM, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
They're two separate issues. It looks like I referenced the wrong Jira in my reply to Michael's comment. The correct Jira for this issue, involving the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>. Oops. I'll retest shortly.
Agreed, the 180-after-183 no audio issue described in FS-7207 <https://freeswitch.org/jira/browse/FS-7207> is clearly different. Sorry for the confusion.
- Jeff
On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian@freeswitch.org> wrote:
If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify?
On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre@gmail.com> wrote:
Just an observation, but what you wrote in the Jira appears to be different from your original email.
'In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.'
So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?
On 16 March 2015 at 14:18, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Michael,
I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.
- Jeff
On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges < m.stgeorges@csl-technologies.com> wrote:
Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
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Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch.
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please follow up on the jira, not here. Thanks Mike
On Mar 17, 2015, at 10:50 AM, Jeff Pyle <jpyle@fidelityvoice.com> wrote:
Confirmed, this issue is still present in FreeSWITCH Version 1.5.15b+git~20150313T215013Z~9872c52c57~64bit (git 9872c52 2015-03-13 21:50:13Z 64bit).
- Jeff
On Mon, Mar 16, 2015 at 11:58 AM, Jeff Pyle <jpyle@fidelityvoice.com <mailto:jpyle@fidelityvoice.com>> wrote: They're two separate issues. It looks like I referenced the wrong Jira in my reply to Michael's comment. The correct Jira for this issue, involving the 200 OK, is FS-7202 <https://freeswitch.org/jira/browse/FS-7202>. Oops. I'll retest shortly.
Agreed, the 180-after-183 no audio issue described in FS-7207 <https://freeswitch.org/jira/browse/FS-7207> is clearly different. Sorry for the confusion.
- Jeff
On Mon, Mar 16, 2015 at 11:47 AM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote: If that is what is going on, then thats a critical detail that wasn't outlined in the JIRA, maybe we can get a new test on the latest code to verify?
On Mon, Mar 16, 2015 at 10:43 AM, Steven Ayre <steveayre@gmail.com <mailto:steveayre@gmail.com>> wrote: Just an observation, but what you wrote in the Jira appears to be different from your original email.
'In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200.'
So it sounded like you heard both ringbacks and nothing after 200. In the Jira you don't talk about the 200 at all, just issues before then. Do you have audio working on receiving 200 now?
On 16 March 2015 at 14:18, Jeff Pyle <jpyle@fidelityvoice.com <mailto:jpyle@fidelityvoice.com>> wrote: Michael,
I don't believe so. I haven't seen any activity in the Jira <https://freeswitch.org/jira/browse/FS-7207>.
- Jeff
On Fri, Mar 13, 2015 at 10:18 PM, Michael St-Georges <m.stgeorges@csl-technologies.com <mailto:m.stgeorges@csl-technologies.com>> wrote: Steven Ayre <steveayre@...> writes:
This sounds like it belongs on Jira so the issue can be tracked.
On 27 January 2015 at 00:04, Jeff Pyle <jpyle-
eOOfO1YW0K2EOSkOl7zanAC/G2K4zDHf@public.gmane.org <mailto:G2K4zDHf@public.gmane.org>> wrote:
Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit).
The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with.
The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example:
- Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183).
In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio.
I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome!
- Jeff
________________________________________________________________________ _
Professional FreeSWITCH Consulting Services:consulting- YF8E+gPBBv73h3GqohbjpQ@public.gmane.orghttp://www.freeswitchsolutions.co <http://www.freeswitchsolutions.co/> m Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www .cluecon.com <http://cluecon.com/> FreeSWITCH-users mailing listFreeSWITCH-users <at> lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswi <http://lists.freeswitch.org/mailman/listinfo/freeswi> tch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- <http://lists.freeswitch.org/mailman/options/freeswitch-> usershttp://www.freeswitch.org <http://www.freeswitch.org/>
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Was this ever fixed? I have a similar problem with the behavior of a Samsung PBX that connects to a FreeSwitch. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
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participants (5)
-
Brian West
-
Jeff Pyle
-
Michael Jerris
-
Michael St-Georges
-
Steven Ayre