rtpmap question PCMU/PCMA
Hi All, I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer. Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'? I'm using sipdroid on android and it rejects this with 'codec not supported' Thanks o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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Hi David, Its an INVITE. Full invite below: INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr> Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To: <sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous <sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv Regards Mark On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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Ah ok. Are you sure your device is not choking because of the T38-related lines in the SDP ? Until now, all the devices I used only advertise T38 in the Re-INVITE, and I think (to be confirmed) that's the "standard" way to do it. I guess there could be a reason not to advertise it in the initial INVITE. As for the missing rtpmap lines, I don't know if it's valid not to send them. I guess it also relies on the device at the other end. Can you confirm how is this related to FS ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 31/05/2010 à 16:57, Mark Campbell-Smith a écrit :
Hi David,
Its an INVITE. Full invite below:
INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr
Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178 :5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To: <sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous <sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp
v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv
Regards Mark
On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone@gmail.com
wrote: Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp. Mike On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
Hi David,
Its an INVITE. Full invite below:
INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr> Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To: <sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous <sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp
v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv
Regards Mark
On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
Hi, I think its a perfectly reasonable invite, including the shiny new capabilities stuff which should reach full RFC status shortly. As it is new, I think the jury is currently out on whether existing poorly implemented SIP packages will choke on it. Steve On 06/01/2010 12:35 AM, Michael Jerris wrote:
This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
Mike
On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
Hi David,
Its an INVITE. Full invite below:
INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr> Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To:<sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous<sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp
v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv
Regards Mark
On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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Thanks Guys. I should have noted that this is not related to an FS fault at all. When using FS in bypass media mode, the call is rejected by the client. Without bypass media mode it works. I just know there is a huge sip knowledge on this mailing list, and would be able to get my answers easily. I guess the android client sipdroid does not like the broken and/or shiny new SDP format. Cheers On Mon, May 31, 2010 at 7:42 PM, Steve Underwood <steveu@coppice.org> wrote:
Hi,
I think its a perfectly reasonable invite, including the shiny new capabilities stuff which should reach full RFC status shortly. As it is new, I think the jury is currently out on whether existing poorly implemented SIP packages will choke on it.
Steve
On 06/01/2010 12:35 AM, Michael Jerris wrote:
This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
Mike
On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
Hi David,
Its an INVITE. Full invite below:
INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr> Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To:<sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous<sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp
v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv
Regards Mark
On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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Coming back to my original question: Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'? Is it possible to add the end of the SDP parameters by doing something like: <extension name="fix_phonzo" continue="true"> <condition field="caller_id_number" expression="123456"> <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> </condition> </extension> Can this be done anywhere in the dialplan as long as its before the bridge command? On Mon, May 31, 2010 at 9:15 PM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Thanks Guys.
I should have noted that this is not related to an FS fault at all. When using FS in bypass media mode, the call is rejected by the client. Without bypass media mode it works.
I just know there is a huge sip knowledge on this mailing list, and would be able to get my answers easily.
I guess the android client sipdroid does not like the broken and/or shiny new SDP format.
Cheers
On Mon, May 31, 2010 at 7:42 PM, Steve Underwood <steveu@coppice.org> wrote:
Hi,
I think its a perfectly reasonable invite, including the shiny new capabilities stuff which should reach full RFC status shortly. As it is new, I think the jury is currently out on whether existing poorly implemented SIP packages will choke on it.
Steve
On 06/01/2010 12:35 AM, Michael Jerris wrote:
This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
Mike
On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
Hi David,
Its an INVITE. Full invite below:
INVITE sip:gw+Phonzo@124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0 Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr> Via: SIP/2.0/UDP 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0 Via: SIP/2.0/UDP 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061 Max-Forwards: 16 From: 010711xxxx <sip:711xxxx@80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco To:<sip:4610xxxxxxx@80.232.37.178> Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE. CSeq: 200 INVITE Contact: Anonymous<sip:80.232.37.178:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 1658214937-1822691807-2631794736-96895450 h323-conf-id: 1658214937-1822691807-2631794736-96895450 Content-disposition: session Content-Length: 364 Content-Type: application/sdp
v=0 o=Sippy 141730476 0 IN IP4 80.232.37.178 s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 a=sendrecv
Regards Mark
On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone@gmail.com> wrote:
Mark, This looks like a T38 Re-INVITE, but a weird one. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
Le 31/05/2010 à 16:12, Mark Campbell-Smith a écrit :
Hi All,
I'm sure I've discussed this before, but I searched through my gmail and google and couldn't find the answer.
Below is the SDP parameters from my sip provider. Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
I'm using sipdroid on android and it rejects this with 'codec not supported'
Thanks
o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx s=- t=0 0 m=audio 47676 RTP/AVP 8 0 18 101 c=IN IP4 xxx.xxx.xxx.xxx a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 ]
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No it is NOT mandatory to include them. Only when they are on defined codec numbers. /b On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote:
Coming back to my original question: Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
Is it possible to add the end of the SDP parameters by doing something like: <extension name="fix_phonzo" continue="true"> <condition field="caller_id_number" expression="123456"> <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> </condition> </extension>
Can this be done anywhere in the dialplan as long as its before the bridge command?
Thanks Brian. I didnt think it was mandatory. As my client is not accepting this sdp format, can I change it? I've tried with <action application="export" data="codec_string=PCMA,PCMU"/> and <action application="export" data="absolute_codec_string=PCMA,PCMU"/> and <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> and have late negotiation active, but nothing is changed in the SDP. Basically I want change v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400 to something like: v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 Is there an 'easy' way to do this? Thanks On Wed, Jun 2, 2010 at 4:21 PM, Brian West <brian@freeswitch.org> wrote:
No it is NOT mandatory to include them. Only when they are on defined codec numbers.
/b
On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote:
Coming back to my original question: Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
Is it possible to add the end of the SDP parameters by doing something like: <extension name="fix_phonzo" continue="true"> <condition field="caller_id_number" expression="123456"> <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> </condition> </extension>
Can this be done anywhere in the dialplan as long as its before the bridge command?
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Ahha... I found why it wasn't working. I have media_bypass active and it appears this does not work when media_bypass is active. Is there any reason for this? Why can't FreeSWITCH modify the signalling and then let the users negotiate which codec to use? If I modify the codec list to be one that is not supported by the server, then its my fault. On Wed, Jun 2, 2010 at 5:03 PM, Mark Campbell-Smith <mcampbellsmith@gmail.com> wrote:
Thanks Brian. I didnt think it was mandatory. As my client is not accepting this sdp format, can I change it?
I've tried with <action application="export" data="codec_string=PCMA,PCMU"/> and <action application="export" data="absolute_codec_string=PCMA,PCMU"/> and <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/>
and have late negotiation active, but nothing is changed in the SDP.
Basically I want change v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 audio RTP/AVP 8 a=cdsc: 2 image udptl t38 a=cpar: a=T38FaxUdpEC:t38UDPRedundancy a=cpar: a=T38FaxVersion:0 a=cpar: a=T38MaxBitRate:14400
to something like: v=0 o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178 s=- t=0 0 m=audio 36824 RTP/AVP 8 0 18 101 c=IN IP4 213.50.91.3 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000
Is there an 'easy' way to do this? Thanks
On Wed, Jun 2, 2010 at 4:21 PM, Brian West <brian@freeswitch.org> wrote:
No it is NOT mandatory to include them. Only when they are on defined codec numbers.
/b
On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote:
Coming back to my original question: Is it mandatory to always include the rtpmap details for PCMU/PCMA codes? For example something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
Is it possible to add the end of the SDP parameters by doing something like: <extension name="fix_phonzo" continue="true"> <condition field="caller_id_number" expression="123456"> <action application="export" data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/> <action application="export" data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/> </condition> </extension>
Can this be done anywhere in the dialplan as long as its before the bridge command?
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participants (5)
-
Brian West
-
David Ponzone
-
Mark Campbell-Smith
-
Michael Jerris
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Steve Underwood